Re: [Other] VOIP.MS in VOIP Tech Chat

Re: [Other] VOIP.MS

Sat, 18 Jul 2009 23:39:17 EDT

PX Eliezer : No, no.

That's a totally different page.

Voip.MS has TWO pages called Features.

Here's the content that Mango2 was talking about:

------------------------------------

On this page, we list feature requests and suggestions from both our customers and staff. Below you can find the suggested features, list of accepted features and list of in-development features.

In Development and/or Testing

* Failover modified to allow Call Routing on Call Status (No Answer, Busy, Unreachable)
* CallerID Name Prefix per DID
* TF Value/Premium routing on a separate setting / Free Value TF Calls
* New website (Not interface, the main website)

Accepted / Planned for Development

* DISA (Direct Inward Dialing). Termination access via a DID that doesn't require Callback (Calling card style)
* Call Back routing according to CallerID + PINs
* Feature Codes
* Line Hunting
* An Option for IVRs that make unanswered calls to subaccounts end up in subaccount's voicemail instead of main DID Voicemail
* Local Access Numbers in major cities
* TF Value/Premium routing on a separate setting / Free Value TF Calls
* Credit Card Auto-replenishment
* Virtual Fax (Web Fax)
* Click to Call and Web Call back
* Better Customer Portal Documentation
* More Configuration Samples for SIP devices and softphones
* Way to navigate the interface the interface with mobiles or other devices that don't support Javascript

Latest Completed Suggestions and Requests

* CallerID Filtering
* CallerID Filtering Wildcards
* Ring Groups
* Calling Queues
* IVR (Digital Receptionist)
* Time Conditions
* User Recordings
* Virtual Numbers (Incoming SIP URI)
* Printable Invoices
* Premium Toll-free
* Better Canadian Pricing
* Better Quebec / Ontario DID Coverage

Re: [Other] VOIP.MS

Sat, 18 Jul 2009 23:18:54 EDT

garys_2k : You don't HAVE to log in. I went to the TOS page (available at the log in page), then "Home," then to Features. A bit roundabout, but no need to have an account. I have no idea why the direct link won't work unless it's from their home page, that is a dumb idea (most likely a mistake).

Re: [Other] VOIP.MS

Sat, 18 Jul 2009 23:10:39 EDT

PX Eliezer :

said by Mango :

I just noticed this page for the first time. Has it always been there and I'm just not observant? Or is it new?

»https://www.voip.ms/m/features.php

m. :)

I sure as hell never noticed it, and I thought that I had fully, carefully, and aggressively, explored their website.

The page content is certainly impressive!!

------------------------------

HA!

They buried the link!

You have to login first, then pulldown the Support menu. I have not needed Support lately, so never saw the link to that page.

It should be more prominently featured, not hidden.

Reminds me of this:

One hundred years ago, in the time before Dick Cheney, US Vice Presidents were very unimportant.

Hence this story told by Thomas R. Marshall (Woodrow Wilson's VP): "Once there were 2 brothers. One ran off to sea, the other became Vice-President. Neither were heard from again".

Re: [Other] VOIP.MS

Sat, 18 Jul 2009 22:56:33 EDT

Mango : I just noticed this page for the first time. Has it always been there and I'm just not observant? Or is it new?
»https://www.voip.ms/m/features.php

m. :)

Re: [Other] VOIP.MS

Fri, 17 Jul 2009 17:46:51 EDT

espaeth :

said by kenja00 :

Wierd... Seems I can't make premium calls right now. I can force it to dial value or I can dial regular -- but premium calls are just getting downgraded... I guess it's saving me money, but I'd like to keep my callers on the highest quality possible...

I made a few test calls today and I'm seeing the same thing. My account is set to use the premium routes, but all calls are showing up in the CDR as value-rate calls.

Re: [Other] VOIP.MS

Tue, 14 Jul 2009 10:37:46 EDT

priller :

said by apn :

Looking online, I see the following listed;

Reg Min Expires: 1 (default)
Reg Max Expires: 7200 (default)
Reg Retry Interval: 30 (default)
Reg Max Expires : 1200 (default)

Which suggests to me that;

Registration is good for a maximum of 7200 secs = 2 hours and the box will keep trying every 30 secs, for a total duration of 1200 secs i.e. 40 times before giving up.

We don't have that level of control with the PAP2T, but it seems that the Reg Max Expires is the parameter that should be reduced to 300 secs (5min).

No, no, no.

Reg Max Expires is the maximum registration expiration time accepted from the proxy. The proxy can override what time you have set under "Register Expires". This the longest time that the ATA will accept during the initial registration negotiation. It is not the variable that actually sets the registration interval.

If you want to change your registration to 300 seconds, "Register Expires" is the correct field. voip.ms allows that interval and will not override it.

HIGHLY suggested you do not screw around with the timers you referenced.

And yes, the PAP2T does have this level of control under the Admin - Advanced View SIP config page.

The Admin Guide that accurately describes these timer is available here: »www.cisco.com/en/US/docs/voice_i···-WEB.pdf

Re: [Other] VOIP.MS

Tue, 14 Jul 2009 09:35:22 EDT

apn : I don't have that box, but isn't the Reg timer on the SIP tab under SIP Timer Values (sec)?

Looking online, I see the following listed;

Reg Min Expires: 1 (default)
Reg Max Expires: 7200 (default)
Reg Retry Interval: 30 (default)
Reg Retry Long Interval: 1200 (default)

Which suggests to me that;

Registration is good for a maximum of 7200 secs = 2 hours and the box will keep trying every 30 secs, for a total duration of 1200 secs i.e. 40 times before giving up.

We don't have that level of control with the PAP2T, but it seems that the Reg Max Expires is the parameter that should be reduced to 300 secs (5min).

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 17:58:45 EDT

WildChild : I guess a timeout of 300 secs, or a bit under should be fine on your ATA. I also have a SPA2102.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 13:14:51 EDT

hockeynomad :

said by WildChild :

On my router, the default timeout for UDP connections is 300 secs. I left it at this value and I set the register timeout of my ATA to 180 secs, just to be sure my ATA re-register before the router closes the hole. This 300 secs value on my router can be configured as I use the Tomato WRT firmware on a WRT54G. Usually, default firmwares don't allow the user to change this value. But you have to make sure the registration timeout of the ATA is <= to the timeout of the router.

Unfortunately I this 2102 I can't even find a timeout on the router

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 12:28:37 EDT

WildChild : On my router, the default timeout for UDP connections is 300 secs. I left it at this value and I set the register timeout of my ATA to 180 secs, just to be sure my ATA re-register before the router closes the hole. This 300 secs value on my router can be configured as I use the Tomato WRT firmware on a WRT54G. Usually, default firmwares don't allow the user to change this value. But you have to make sure the registration timeout of the ATA is <= to the timeout of the router. When it's configured this way, there's no need to forward any port on the router. As soon as the ATA initiate the connection, the router will leave the hole opened for incoming packets until the timeout is reached after the last packet has been seen. Also, about registration, I found Toronto's server to be more stable than Montreal's server. I live in Québec City but I get a ping between 20ms and 24ms to the Toronto server.

said by hockeynomad :

Hot in Vancouver? well its a dry hot. Been coldest summer I've seen in Toronto area.

OK, just changed registration expires from 600 to 300.

BTW, I've noticed that the internet light on my ATA has been fluttering the last two weeks. Again a case for bypassing my inside wiring.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 10:44:47 EDT

wysiwyg1972 :

said by Mango :

Set up "Internal Extension Voicemail" on the "Manage Sub-Accounts" page.

OK, I didn't realize this was necessary! I just want to the second phone to see voicemail messages. I don't want to a different mailbox for the sub-account.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 08:24:14 EDT

hockeynomad : Hot in Vancouver? well its a dry hot. Been coldest summer I've seen in Toronto area.

OK, just changed registration expires from 600 to 300.

BTW, I've noticed that the internet light on my ATA has been fluttering the last two weeks. Again a case for bypassing my inside wiring.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 08:14:47 EDT

Mango : Heh - it's too hot to sleep. In Vancouver. I just got back from Alberta and it was seven degrees. The weather has been very backwards lately.

It would certainly be nice if VoIP.ms were able to solve the problem. While you're waiting, you may want to try a few things yourself too. Part of the reason BYOD is less expensive is that sometimes you need to troubleshoot things on your own. Let me know if my suggestions worked, or not.

Registration interval is sometimes known as "Register expires:" and may be found on the Line tab of Sipuraesque devices.

m.
--
Mango's recommended PAP2T settings: »www.toao.net/25/linksys-pap2t-vo···-review/
Linksys/Cisco dial plan tips and tricks: »www.toao.net/108/cisco-dial-plan···-tricks/

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 07:46:22 EDT

hockeynomad : Mango, you're up early. the registration interval? on ATA device.

I really want voip.ms support to follow through on this and offer a potential cause or at least a few.

I've had inside wiring issues and maybe a short, which hopefully a n alternate CAT5 from the basement near the demarc point could resolve.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 07:34:33 EDT

Mango : What if you create a CallerID filter that sends those specific callers to your phone?

Edit: Or just wildcard all callers to your phone.

Also, I'm sure you've done the usual dance: turned on NAT in the VoIP.ms control panel and your device and set your registration interval to 300 seconds or less.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 07:32:13 EDT

hockeynomad :

said by Mango :

said by hockeynomad :

Does anyone else have this problem with incoming calls unable to connect?

Specific callers cannot get through and some go directly to voice mail. I can see a CDR though.

This started only after CallerID filtering came about.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 07:04:52 EDT

anon : No problems here. The service runs like grease through a goose.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 06:29:48 EDT

sream : Haven't had a single issue since I started using them.

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 06:15:44 EDT

Mango :

said by hockeynomad :

Does anyone else have this problem with incoming calls unable to connect?

I'm late to the party here ;) But I do not have this problem.

I'm not clear what happens to the incoming calls. Do they get sent to Voicemail, or something else? Does it still only happen for specific calls or all calls? Is there any pattern to the calls it happens with?

m.
--
Mango's recommended PAP2T settings: »www.toao.net/25/linksys-pap2t-vo···-review/
Linksys/Cisco dial plan tips and tricks: »www.toao.net/108/cisco-dial-plan···-tricks/

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 00:05:11 EDT

kenja00 : Wierd... Seems I can't make premium calls right now. I can force it to dial value or I can dial regular -- but premium calls are just getting downgraded... I guess it's saving me money, but I'd like to keep my callers on the highest quality possible...

Re: [Other] VOIP.MS

Sun, 12 Jul 2009 21:30:55 EDT

hockeynomad : Does anyone else have this problem with incoming calls unable to connect?

At this point, Callcentric appears really tempting.

Re: [Other] VOIP.MS

Sun, 12 Jul 2009 21:22:57 EDT

Mango : Set up "Internal Extension Voicemail" on the "Manage Sub-Accounts" page.

Re: [Other] VOIP.MS

Sun, 12 Jul 2009 19:54:26 EDT

wysiwyg1972 : I have a SIP phone (Aastra 9143i) which doesn't have the MWI light come on when there's voicemail. It is set up on the secondary account.

When I check the portal for the secondary account, it claims that I can change voicemail indicator settings there, but there's nothing for that on the page. The "main account" is set with MWI indicator enabled, but it seems to affect the main account only. (Which I use and works fine on the PAP2T).

Also, it's similar, *97 from the main account takes me directly to "enter password", while on the secondary account, it asks for a login.

Am I missing something?

Re: [Other] VOIP.MS

Sat, 11 Jul 2009 07:52:57 EDT

hockeynomad :

said by PX Eliezer :

Yes.

Here are the comments from their website:

Issue Tracker

Login / Website issue 2009-07-10 11:47:13
[ Status: RESOLVED ]
This morning we experienced a database issue that prevented users from loging in to the customer portal. We have resolved the issue. We apologize for any inconvenience this may have caused.

DID Issue 2009-07-09 20:11:24
[ Status: RESOLVED ]
One of our carriers is experiencing issues affecting some DID Numbers. We're working with them to re-establish service asap for those affected numbers. Thank you for your patience and we apologize for the inconvenience.

.

OK, so now I had the same problem yesterday of an inbound call not connecting. There is a CDR report so it is probably not being blocked. It is an anonymous call and I am certain it is not set to block them.

This sucks as I am in job search mode and calls not coming in is deadly.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 15:48:00 EDT

dalrun : The voip.ms site has been timing out on me for weeks, if not months. Very strange.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 12:08:39 EDT

hockeynomad :

said by shighfield :

Or maybe they did, and screwed something! The important thing is.... the phones work!

That is the important thing! I just can't foward my calls to the cell so am stuck at my desk on a beautiful friday! ;)

Edit: Nevermind, the *72 code works for fowarding! I didn't know that. :)

so to forward calls to another number, use the *72 feature on the phone dial pad, or on the ATA device?

Are there prompts then? and how to disable?

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 11:55:25 EDT

PX Eliezer : Yes.

Here are the comments from their website:

Issue Tracker

Login / Website issue 2009-07-10 11:47:13
[ Status: RESOLVED ]
This morning we experienced a database issue that prevented users from loging in to the customer portal. We have resolved the issue. We apologize for any inconvenience this may have caused.

DID Issue 2009-07-09 20:11:24
[ Status: RESOLVED ]
One of our carriers is experiencing issues affecting some DID Numbers. We're working with them to re-establish service asap for those affected numbers. Thank you for your patience and we apologize for the inconvenience.

.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 11:48:08 EDT

N9MD :

said by N9MD :

Friday 7/10 at 1415Z and still cannot log onto Voip.ms website

They're ba-a-a-a-ck!

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 10:51:27 EDT

sapstar :

said by shighfield :

Or maybe they did, and screwed something! The important thing is.... the phones work!

That is the important thing! I just can't foward my calls to the cell so am stuck at my desk on a beautiful friday! ;)

can't you forward them from your Gateway?

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 10:41:29 EDT

shighfield :

Or maybe they did, and screwed something! The important thing is.... the phones work!

That is the important thing! I just can't foward my calls to the cell so am stuck at my desk on a beautiful friday! ;)

Edit: Nevermind, the *72 code works for fowarding! I didn't know that. :)

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 10:29:30 EDT

sapstar : They must be updating their portal site for their new rate centers.

Or maybe they did, and screwed something! The important thing is.... the phones work!

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 10:16:47 EDT

N9MD :

said by burgerwars :

Anyone else get an HTTP 500 Internal Server Error message when trying to login?

Friday 7/10 at 1415Z and still cannot log onto Voip.ms website ... but calls in and out are working fine, with clear audio.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 10:06:19 EDT

hockeynomad :

said by burgerwars :

Anyone else get an HTTP 500 Internal Server Error message when trying to login?

Yesterday the service was flakey with incoming calls. At least for me.

OK, so I wasn't alone.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 09:29:23 EDT

anon :

said by burgerwars :

Anyone else get an HTTP 500 Internal Server Error message when trying to login?

Yesterday the service was flakey with incoming calls. At least for me.

I can't login either, but service inbound and outbound is fine.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 09:24:15 EDT

burgerwars : Anyone else get an HTTP 500 Internal Server Error message when trying to login?

Yesterday the service was flakey with incoming calls. At least for me.

Re: [Other] VOIP.MS

Fri, 10 Jul 2009 08:46:45 EDT

hockeynomad : I made the adjustments, thanks. I can get thru to that number now.

Now, another number cannot connect to me. :(

I may have to go the premium route,
and the voip.ms site is down now.

Re: [Other] VOIP.MS

Thu, 09 Jul 2009 07:38:03 EDT

priller :

said by papaskitch :

said by hockeynomad :

I can login and do that, but how can I force the G729 on the Linksys SPA2102 device?

The SPA2102 should fall back to G.729 if you deselect G.711 in the voip.ms portal.

Correct. Both ends will negotiate a compatible codec.

If you want to enforce it from the SPA end. You need to go to Voice --> Line X --> Audio Configuration and make G.729 your Preferred Codec. You can also enable/disable codecs or enforce just using the preferred.

»ui.linksys.com/files/SIPURA/SPA-···oice.htm

Re: [Other] VOIP.MS

Thu, 09 Jul 2009 00:48:46 EDT

trev :

said by hockeynomad :

but how can I force the G729 on the Linksys SPA2102 device?

I don't know how voip.ms has your device configured, but the default setting to force G729a on a Sipura/Linksys device is to dial *02729, at which point you'll have dial tone and can dial your call as you usually would. This will force G729 and the call will fail rather than use any other codec.

To prefer G729a, dial *01729 and other codecs will be used as a fallback if it can't be negotiated.

Re: [Other] VOIP.MS

Wed, 08 Jul 2009 19:19:53 EDT

papaskitch :

said by hockeynomad :

I can login and do that, but how can I force the G729 on the Linksys SPA2102 device?

The SPA2102 should fall back to G.729 if you deselect G.711 in the voip.ms portal.

said by hockeynomad :

Also I am suddenly experiencing dropped calls. Using the premium service, is that less likely?

The premium routes are less likely to suffer dropped calls or other issues than the value routes. I can't use the value routes as the connections are usually flakey (and CID doesn't usually get passed). Of course, YMMV.

Re: [Other] VOIP.MS

Wed, 08 Jul 2009 17:08:27 EDT

hockeynomad : Regarding my issue with one way calling and the other party cannot hear me; voip.ms support suggested I only use the G729 codec on the customer portal.

I can login and do that, but how can I force the G729 on the Linksys SPA2102 device?

Also I am suddenly experiencing dropped calls. Using the premium service, is that less likely?

Re: [Other] VOIP.MS

Sat, 20 Jun 2009 11:01:52 EDT

papaskitch : Well, I contacted support, and the responded a day later saying they would send the LNP through a different carrier, and less than a few days later, I have an FOC of June 30.

I hope all goes well.

Re: [Other] VOIP.MS

Sat, 13 Jun 2009 12:22:57 EDT

hockeynomad : I had inside wiring issues (again) and once that was supposedly taken care of, I am encountering issues I rarely had: dropped calls, calls going directly to voicemail, an outgoing call the other party couldn't hear anything, etc.

could this again be related to wiring?

One last option I am considering is placing my modem in the basement, ie: as close as possible to the demarc point and installing a CAT5 wiring to bypass the internal wiring to my computer.

I spent an hour with chat support about the voicemail issue and they are "escalating it".

The CAT5 should take out the internal wiring issue?

Re: [Other] VOIP.MS

Sat, 13 Jun 2009 03:54:38 EDT

Mango : SWEET! I am very glad to hear it :)

Re: [Other] VOIP.MS

Fri, 12 Jun 2009 18:02:11 EDT

WildChild : My number was ported today! I'm still within the 30 days delay I'm supposed to have with Videotron. Everything seems to work fine right now. I hope it will stay like this! ^_^

said by WildChild :

Some update about my complicated LNP... I got more information from Videotron. They told me they never received any request for my number, but that it can happen when the LNP form isn't filled correctly by the requesting carrier. They can say anything but I'm interested to know... Who manages LNP requests between carriers? Do they contact each others directly of there is an entity in the middle?

Also, they told me that the LNP had to be done after the end of my contract (May 16th), after what the contract will be renewed for 1 year BUT, there is 30 days of delay after the renewal of the contract to transfer the number without any penalty for breach of contract. In the end, it means that voip.ms has until June 15th to transfer the number.

I'll keep you updated!

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 20:54:25 EDT

hockeynomad :

said by emoci :

It seems you're on interleave which adds some latency...plus the fact that the line has partial Sync both up and down is not right....has your line always looked this way....

said by hockeynomad :

Been getting complaints on inbound calls that a high pitched sound is heard.

Well hear are my stats, if anyone can make any sense of them.

Are you still with Teksavvy...post in the direct forum...
-------------------------------------
Here is the response from Ts:

Here are the current line stats.

Line Status: In Service UpTime:
Line Profile: d2496/u640 on Interleaved Last State Change: Thu Jun 11 13:15:15 EDT 2009
Operational Status Speed (Kbs) Relative Capacity Occupation (%) Noise Margin (0..31 dB) Signal Power (0..20 dBm) Attenuation (0..60 dB) Block count
UpStream 640 66 18.0 12.0 30.0 6.007564E7
DownStream 2496 68 15.0 18.0 51.0 6.1453888E7

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 20:13:59 EDT

sapstar :

said by WildChild :

Who is your current carrier?

Bell

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 20:06:31 EDT

WildChild : Who is your current carrier?

said by sapstar :

Well, not to try to impress you or anything, but...

My port is now 10 weeks old. Documents were sent to carrier on April 2nd. I'm contacting support on a weekly basis. Keep reading that "soon" word. Seems the carrier is totally ignoring us.

I even contacted MartinM directly. I guess he's getting fed up as well, since he's not even replying to me anymore.

My hope is now that I'm leaving on monday for a 3-week vacation and when I'm back, my port will have been fully completed.

What?, I can dream, can't I ?

said by papaskitch :

said by db2222 :

It took more then 8 weeks to port a number to voip.ms, but it was ported in the end. It isn't their fault.

Wow that's a long time. I thought the process was getting better. I'm not blaming Voip.ms, in fact, they shouldn't have to pay their carrier for the LNP at this point (or they should at least be getting a discount which they could pass on to me :p).

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 16:23:04 EDT

apn : 4wks isn't that long, but I've had relatives port in 15 business days.

Not aware of a "policy" on refunding LNP requests, but I got one.

Port request was submitted on Feb28 and included a letter indicating my current service was ending Apr22. I waited, waited and no FOC came through.

I started to get worried and engaged voip.ms on Apr20 who said they were waiting on my current carrier. It was two days until I lose service, so I called my carrier who indicated that they'd not received an LNP request. WTH? :mad:

Then I politely tore a strip off the voip.ms rep, demanding that they resend the request and that I'd call my carrier to confirm receipt by end of day.

The request was indeed received (this time) and FOC completed on Apr22. In apology, Stephen offered me a refund, which I accepted.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 15:42:02 EDT

hockeynomad : I gave the stats to TS Direct and live chat support with voip.ms they advised I change NAT settings from NO to YES. The settings save now.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 14:45:34 EDT

emoci :

said by hockeynomad :

Been getting complaints on inbound calls that a high pitched sound is heard.

Well hear are my stats, if anyone can make any sense of them.

Are you still with Teksavvy...post in the direct forum...

It seems you're on interleave which adds some latency...plus the fact that the line has partial Sync both up and down is not right....has your line always looked this way....

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 14:29:44 EDT

Mango :

said by hockeynomad :

I tried that, but it won't save. Interesting.

That's...curious. Check the Provision tab and be sure that Provision Enable is set to "no".

Does the high-pitched sound only happen when people call you, not when you call people? If so it is unlikely the issue is to do with your equipment.

said by sapstar :

My port is now 10 weeks old.

Heh. You beat me - I was nine weeks two days.

m.
--
Mango's recommended PAP2T settings: »www.toao.net/25/linksys-pap2t-vo···-review/
Linksys/Cisco dial plan tips and tricks: »www.toao.net/108/cisco-dial-plan···-tricks/

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 13:26:06 EDT

hockeynomad : Been getting complaints on inbound calls that a high pitched sound is heard.

Well hear are my stats, if anyone can make any sense of them.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 09:35:54 EDT

sapstar : Well, not to try to impress you or anything, but...

My port is now 10 weeks old. Documents were sent to carrier on April 2nd. I'm contacting support on a weekly basis. Keep reading that "soon" word. Seems the carrier is totally ignoring us.

I even contacted MartinM directly. I guess he's getting fed up as well, since he's not even replying to me anymore.

My hope is now that I'm leaving on monday for a 3-week vacation and when I'm back, my port will have been fully completed.

What?, I can dream, can't I ?

said by papaskitch :

said by db2222 :

It took more then 8 weeks to port a number to voip.ms, but it was ported in the end. It isn't their fault.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 09:28:28 EDT

anon :

I tried that, but it won't save. Interesting.

That *is* interesting. Sounds like it may be a clue, but if it is, I can't follow it.

the proxy fallback Intvl is also set at 3600. Should that be changed as well?

Beats me. But I didn't change it on mine and yet everything is now copacetic.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 09:18:18 EDT

hockeynomad :

said by apn :

As mentioned in the other forum (;)), it might be de/registration.

If you're using a PAP2T, there are Make/Ans calls w/o Registration settings in the LINE(x) tabs. Both are defaulted to OFF, but enabling the Ans setting might resolve your problem.

It worked okay till now. Why could it be causing trouble now.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 09:17:25 EDT

hockeynomad : I tried that, but it won't save. Interesting.

the proxy fallback Intvl is also set at 3600. Should that be changed as well?

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 08:41:43 EDT

papaskitch :

said by db2222 :

It took more then 8 weeks to port a number to voip.ms, but it was ported in the end. It isn't their fault.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 08:01:26 EDT

anon : I had the same problem: messages were going to voicemail when they shouldn't. It turned out that the registration was dropping. I fixed it by changing 'register expires' from the default of 3600 to 600.

This setting is on each of the 2 'Line' tabs of my pap2.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 07:57:04 EDT

anon : It took more then 8 weeks to port a number to voip.ms, but it was ported in the end. It isn't their fault.

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 07:51:45 EDT

papaskitch : I've been waiting for an LNP for over 4 weeks now. Just wondering what kind of luck people have had getting a refund, or what the policy is for refunds on LNPs.

Re: [Other] VOIP.MS

Wed, 10 Jun 2009 20:57:38 EDT

Mango : Check the following:

- Be sure that Failover is set to 'none'.
- Be sure that Voice Mail is also set to 'none'.
- Set the Dial Time Out to some large number, more than the number of rings before your answering machine kicks in.
- If you use a sub account, be sure that Internal Extension VoiceMail is also set to 'none'.
- Same deal for Internal Extension Ringing Time as Dial Time Out.
- Be sure no forwarding in your ATA is accidentally sending callers to voicemail.
- Be sure no Caller ID Filtering is accidentally sending callers to voicemail.
- Be sure Register Expires (Line tab of Sipuraesque ATAs) is set to to something low like 300 or even 120.

Finally, are you able to duplicate the issue? It sounds like some calls are being sent to voicemail and some go to your answering machine. Is there any pattern to this?

If you've considered all of this and the issue still occurs, then you should open a ticket with VoIP.ms support. Include the specific calls in which the issue happened (check the CDR) if the issue does not happen on every call.

Please post back when you get the problem sorted so that future customers with the same problem will be able to solve it, too :)

m.
--
Mango's recommended PAP2T settings: »www.toao.net/25/linksys-pap2t-vo···-review/
Linksys/Cisco dial plan tips and tricks: »www.toao.net/108/cisco-dial-plan···-tricks/

Re: [Other] VOIP.MS

Wed, 10 Jun 2009 20:11:41 EDT

apn : As mentioned in the other forum (;)), it might be de/registration.

If you're using a PAP2T, there are Make/Ans calls w/o Registration settings in the LINE(x) tabs. Both are defaulted to OFF, but enabling the Ans setting might resolve your problem.

Re: [Other] VOIP.MS

Wed, 10 Jun 2009 15:49:38 EDT

hockeynomad : I do not have the voicemail system of voip.ms enabled so my phone with built in answer machine is handling the messages. for some reason I like to see the number of messages on the box.

The trouble is some calls are routed to the voip.ms voice system and not sure why.

Any ideas?

P.S. I don't want to waste my time in chat support.

Re: [Other] VOIP.MS

Tue, 26 May 2009 21:34:04 EDT

burgerwars : Thanks again.

Re: [Other] VOIP.MS

Tue, 26 May 2009 15:22:30 EDT

MartinM :

said by burgerwars :

Thanks anyway. I guess I can use the "poor man's solution" by telling whoever is calling me in my recorded IVR announcement what extension numbers to press. Also, your website is down right now. Have any idea when it will be back up?

I've spoken with a dev and it should not be too hard to implement. We'll try to include it on our revamp of the DID Edit page.

The portal is backup, was an emergency maintenance as indicated on our website.

Re: [Other] VOIP.MS

Tue, 26 May 2009 14:53:50 EDT

burgerwars : Thanks anyway. I guess I can use the "poor man's solution" by telling whoever is calling me in my recorded IVR announcement what extension numbers to press. Also, your website is down right now. Have any idea when it will be back up?

Re: [Other] VOIP.MS

Tue, 26 May 2009 14:50:01 EDT

usa2k : If you make your own phone pause ...

(Poor man's solution)
Send the Voip.ms part |pause| send digits assuming Voip.ms connected.

Re: [Other] VOIP.MS

Tue, 26 May 2009 14:39:23 EDT

MartinM :

said by burgerwars :

With VOIP.MS in the Call Forwarding section, is there any way to place a pause, wait for the forward to connect, and then automatically put an extension number in? I saw in an earlier post a mention of addings P's for pause in another option, but it doesn't work here (just dead air). Any suggestions?

This is not possible because the number you have configured there, are not DTMF tones, simply the number we send to the PSTN.

Pausing and then dialing would involve a totally different process which consist of waiting for a call to be answered, then send DTMF Tones for the proper extension. Unfortunatly, that's not something we support at this time but I did send it to our big piles of suggestions. Although many suggestions from our customers make it to the live interface, I can't promise anything at this time.

Re: [Other] VOIP.MS

Mon, 25 May 2009 21:43:51 EDT

sream : pauses for extensions, etc have always been a pain be it on speed dial entries , cell phone contacts, etc.

Re: [Other] VOIP.MS

Mon, 25 May 2009 21:35:24 EDT

soitgoes2 :

That would be a cool feature, but I don't know of any way to do it with voip.ms or any other provider. Maybe it's possible. Voip.ms has virtually zero documentation.

Re: [Other] VOIP.MS

Mon, 25 May 2009 21:32:45 EDT

burgerwars : With VOIP.MS in the Call Forwarding section, is there any way to place a pause, wait for the forward to connect, and then automatically put an extension number in? I saw in an earlier post a mention of addings P's for pause in another option, but it doesn't work here (just dead air). Any suggestions?

Re: [Other] VOIP.MS

Mon, 25 May 2009 17:10:31 EDT

sream : Any possibility on some documentation for the reseller interface and how it relates to subaccounts? I'm confused as to how much configuration control someone will have if they are added under the reseller section.

Re: [Other] VOIP.MS

Sun, 24 May 2009 16:08:03 EDT

Mango : This issue has been fixed. Thanks Steve :D

Re: [Other] VOIP.MS

Sat, 23 May 2009 18:00:20 EDT

Mango : I want to route a DID to a URI.

EDIT: It appears to be working with one of my DIDs and not the other. I suspect it is because they are on different servers. I need to go out for a few hours but I will confirm this when I get back.

EDIT: The issue only occurs on the server I use (US3B). Go figure :)

m.

Re: [Other] VOIP.MS

Sat, 23 May 2009 17:55:30 EDT

sream : Mango, I Have all sorts of uri stuff going on with voip.ms and would love to help. The problem is that I cannot for the life of me figure out what you're trying to do. Please be more clear.

Re: [Other] VOIP.MS

Sat, 23 May 2009 01:21:35 EDT

Mango : I was just using that URI for testing purposes. I tried another URI from a different provider and was not able to make it work either. Am I doing something wrong?

Edit: typo.

Re: [Other] VOIP.MS

Fri, 22 May 2009 21:49:23 EDT

MartinM : I saw the ticket and I'm not sure I understand why you want to route a VoIP.ms DID to a VoIP.ms URI. Just point the DID to you SIP device it will achieve the same result.

Re: [Other] VOIP.MS

Fri, 22 May 2009 21:23:06 EDT

Mango :

said by MartinM :

Section #2 is used for routing calls to SIP URI's of other companies or your own PBX. For example, if you want to forward your VoIP.ms Number to your Call Centric SIP URI, you create a SIP URI In that section. You then edit your DID and point the routing to that address. When people call your DID, it will ring the SIP Address.

I tried to use this today, both routing the DID directly, and by using CallerID Filtering, but I got a reorder tone when I called my DID. I verified with a softphone that the SIP URI I was using was valid, and I tried another SIP URI just to be absolutely sure.

I reported the issue to Live Chat, and after about half an hour of chatting the tech escalated the issue.

m.
--
Mango's recommended PAP2T settings: »www.toao.net/25/linksys-pap2t-vo···-review/
Linksys/Cisco dial plan tips and tricks: »www.toao.net/108/cisco-dial-plan···-tricks/

Re: [Other] VOIP.MS

Fri, 22 May 2009 18:54:46 EDT

Mango : Awesome! MartinM, can you assure us that this will happen?

Edit: WildChild, I can't remember if you said if your porting fee was refunded, but you may want to ask about this given the time you've spent on this issue. I know VoIP.ms has refunded the porting fee for troublesome ports in the past.

m.

Re: [Other] VOIP.MS

Fri, 22 May 2009 17:01:44 EDT

WildChild : Some update about my complicated LNP... I got more information from Videotron. They told me they never received any request for my number, but that it can happen when the LNP form isn't filled correctly by the requesting carrier. They can say anything but I'm interested to know... Who manages LNP requests between carriers? Do they contact each others directly of there is an entity in the middle?

Also, they told me that the LNP had to be done after the end of my contract (May 16th), after what the contract will be renewed for 1 year BUT, there is 30 days of delay after the renewal of the contract to transfer the number without any penalty for breach of contract. In the end, it means that voip.ms has until June 15th to transfer the number.

I'll keep you updated!

Re: [Other] VOIP.MS

Fri, 22 May 2009 16:24:05 EDT

espaeth : I had occasional problems with DTMF pass-through and accurate CID delivery on TF calls via voip.ms. I haven't used them for TF since 2008, so maybe things have improved, but IMO if they are going to charge for TF termination then it better be flawless.

I find it ironic that the companies I use who provide truly free TF termination do a significantly better job that most of those who charge.

Re: [Other] VOIP.MS

Fri, 22 May 2009 15:54:13 EDT

usa2k : Then it does matter, for you.

As long as you get don't less quality using other means, or waste personal time maintaining another method to dial, or complicate things for others (Especially the wife) :)

I'm not using this service, but I bet I have 2 hours toll free a year.
--

Jim, VoIP since 12/2002, VOIPo 2/7/2007
FAH-Tool ... Whales ... USA2K site ... Artist-247

Re: [Other] VOIP.MS

Fri, 22 May 2009 15:43:09 EDT

Eat Me :

said by usa2k :

$0.15/hr is hardly devastating though.

I make on average 2 hours of TF calls per day at the office. This is stuff like webex and conference calls. If I get the opportunity to work from home (may/may not happen) and I do 2 hours x 20 days that's 6 bucks of making "toll free" calls.

Re: [Other] VOIP.MS

Fri, 22 May 2009 15:34:46 EDT

usa2k : $0.15/hr is hardly devastating though.

Re: [Other] VOIP.MS

Fri, 22 May 2009 15:10:13 EDT

Serbtastic : You can even use the free version of Skype to make TF calls. I use it all the time and it works fine.

Re: [Other] VOIP.MS

Fri, 22 May 2009 15:08:48 EDT

Eat Me :

said by wuliu7 :

Anybody knows whether outbound toll free calls are free at voip.ms?

So many places for free TF calls, including Future Nine and SIPBroker.

In fact you don't even need an account to make TF calls at F9.

Re: [Other] VOIP.MS

Fri, 22 May 2009 14:58:45 EDT

Boolah : No - they cost $0.0025 per minute (1/4 cent per minute). Here's the link (look under the Call Termination heading):

»https://www.voip.ms/specifications.php

Re: [Other] VOIP.MS

Fri, 22 May 2009 14:55:02 EDT

wuliu7 : Anybody knows whether outbound toll free calls are free at voip.ms?

Re: [Other] VOIP.MS

Thu, 21 May 2009 17:41:34 EDT

MartinM :

said by sream :

said by MartinM :

We're currently building a knowledge base that will contain around 1 thousand entries. This should cover most of the more advanced sections of our website and all the technical features.

Hopefully soon. There seems to be a fair amount of confusion regarding the incoming sip uri and the sip uri's used for did routing.

Surprisingly we receive very few requests about this. It's quite straight forward. SIP URI's are used as destinations for your numbers. Virtual Numbers are used for origination via SIP.

Re: [Other] VOIP.MS

Thu, 21 May 2009 17:37:48 EDT

sream :

Hopefully soon. There seems to be a fair amount of confusion regarding the incoming sip uri and the sip uri's used for did routing.

Re: Playing with SIP URI's

Wed, 20 May 2009 12:25:48 EDT

MartinM :

said by apn :

Thanks for the clarification, Martin.

I get the SIP URI thing and have confirmed that I can call my DID via a softphone, so that suggests that my PAP2T is the bottleneck here.

Given the statement about not being able to call virtual DID's over the PSTN (makes sense), I suspect that the limitations of my ATA will apply here also; although I've not (yet) tested if the device will dial a hard-coded IP address in lieu of a DNS url/address.

Here's the scenario I'm trying to test: via my voip.ms account, call another voip.ms subscriber via his SIP URI and vice versa.

To call another subscriber, he will need to give you his virtual number, and you just need to dial it. Our virtual numbers all begin with 111 like you would dial any other phone number. There's no need to enter a SIP URI in your softphone. Remember they are not assigned automatically so your friend will have to create his number via the interface.

If you want to do direct peer to peer with the other person, you don't need to use VoIP.ms, however NAT and SIP Peer-to-Peer do not cooperate very well.

So to make it short, if your friend's Virtual Number is 11111111111, just dial 11111111111 to reach him.

Re: [Other] VOIP.MS

Wed, 20 May 2009 12:20:40 EDT

MartinM : Section #1, is our standard Virtual Numbers, they can be reached via SIP from anywhere and anonymously. They are called Virtual Numbers because that's what they are. They are reachable via SIP (Aka: a SIP URI) from outside. There numbers can be configured just like your standard PSTN DIDs so you can use call treatment on them.

Section #2 is used for routing calls to SIP URI's of other companies or your own PBX. For example, if you want to forward your VoIP.ms Number to your Call Centric SIP URI, you create a SIP URI In that section. You then edit your DID and point the routing to that address. When people call your DID, it will ring the SIP Address.

Section #3 you probably don't need it if you don't understand the use. Some customers had specific needs of having a subaccount directly reachable via a SIP URI without involving call treatment. It's pure SIP to SIP.

I agree with the lack of documentation, but then again, if you don't understand what are SIP URI's, you probably have no use for it.

We're currently building a knowledge base that will contain around 1 thousand entries. This should cover most of the more advanced sections of our website and all the technical features.

Re: Playing with SIP URI's

Wed, 20 May 2009 11:46:37 EDT

apn : Thanks for the clarification, Martin.

I get the SIP URI thing and have confirmed that I can call my DID via a softphone, so that suggests that my PAP2T is the bottleneck here.

Given the statement about not being able to call virtual DID's over the PSTN (makes sense), I suspect that the limitations of my ATA will apply here also; although I've not (yet) tested if the device will dial a hard-coded IP address in lieu of a DNS url/address.

Here's the scenario I'm trying to test: via my voip.ms account, call another voip.ms subscriber via his SIP URI and vice versa.

Re: [Other] VOIP.MS

Wed, 20 May 2009 11:04:18 EDT

PX Eliezer : Agree with that. Thanks for expressing it clearly and in such detail.

Re: [Other] VOIP.MS

Wed, 20 May 2009 11:00:53 EDT

anon : The SIP URI sections of the website are conflicting and confusing. Look at these 3 sections:

1) Virtual SIP numbers are similar to standard DID numbers. The major difference is that virtual SIP numbers are not accessible via "PSTN". They can only be reached via "SIP URI" over internet. For example, if you have a DID number with another provider and they support SIP URI Forwarding, you could forward your number to a virtual number at voip.ms just like if it was one of our numbers.
All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits. The final uri will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server. Example SIP URI: 11102198123@sip.us4.voip.ms

2) Another way of routing incoming calls to your DID numbers is by the use of SIP URI's.
To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section".
Example: 1{DID}@Your_IP_address = 12143221234@128.144.122.12
Example: johnsmith@my-uri.com

3) We now accept incoming SIP URI calls from other networks to VoIP.ms sub accounts. The SIP URI will share the "internal extension" code of your sub account. For this example, we'll pretend your 6 digits account code is 111111 and your subaccount has internal extension "2" (102 to be reached internally). The device or softphone using this sub account is registering to the New York server. The external SIP URI for this sub account would be 1111112@sip.us4.voip.ms. There's a small fee of $0.001 (one tenth of a cent per minute) when using SIP URIs. You can verify the current URI for your sub accounts by clicking on "Manage subaccounts" from the Sub Account tab.

The terms Sip, URI, and Virtual seem to be interchangeable. The costs and choice of sip uri number seems to be different in each section. In number 2, why do I have to put in my home ip address?
I believe there is a way to re-write these sections to make everything more clear.

Re: Playing with SIP URI's

Tue, 19 May 2009 23:08:51 EDT

MartinM : I will agree with you that the fact that they give you right away your single SIP URI is more simple for less advanced users than having to activate your Virtual Number(s) manually like you have to do with VoIP.ms.

However, the act of creating a Virtual Number (or sip URI) our sip uri can be achieved under a minute via our portal, you choose your number and click create.

Our virtual numbers are also dialed the same way, you can have as many as you want for you and your customers and our routing options gives more flexibility on what you want your Virtual Number to achieve.

Thank you for the nice comments as well however the customer was simply asking if his account could be reached via his main account code @ voipms server. Our lack of documentation may have misled him to believe that, I just fail to see how the URLs provided were of any help to him.

----
If anyone need hints regarding how to be reached via SIP, let me know or let our support know and they will explain you how to be reached it is very simple.

Re: Playing with SIP URI's

Tue, 19 May 2009 20:07:44 EDT

PX Eliezer :

said by MartinM :

In fact, It's quite simple to use SIP URIs with VoIP.ms.

Just order a Virtual SIP Number and configure it the way you want. The only restriction is that it can not be called via PSTN.

The other type of SIP URI we have is for advanced users. It's direct calls to subaccounts, but you have no control on call flow this way.

Furthermore, we all due respect, I do not believe the URLs you provided will help the user with his questions on SIP URI configuration.

First off, I tried to say some very positive things about Voip.MS: "Voip.MS has many fine features, and many of their new ones are quite interesting."

I still maintain that CallCentric has an easier approach to SIP URI. With them, there is no need to order a separate "Virtual SIP" number, and there is no need to deal with any "questions on SIP URI configuration".

This is CallCentric's approach. It is so simple, I don't see how questions could occur:

"You can be reached (for SIP calling) at: callcentricnumber@in.callcentric.com"

"A specific example would be:
17770000001@in.callcentric.com"

And yes, I still maintain this is an easier approach than the Voip.MS method.

There are SOME areas in which Voip.MS is AHEAD---such as being able to see the selection of available DID's and set them up instantly. Marvelous!!

But no single Voip provider can be the top in every single category.

VOIP.MS

Tue, 19 May 2009 19:42:37 EDT

burgerwars : I'm always a sucker for a good phone number, but have no need this one:

Browse numbers under Meredith, New Hampshire and there's one that ends in "8000." At least it was still there at 4:40 PM PDT today. VOIP.MS lets you keep most DIDs for only 99 cents per month. If anyone wants it, go get it before it's gone!

Re: Playing with SIP URI's

Tue, 19 May 2009 11:05:11 EDT

MartinM :

said by PX Eliezer :

Voip.MS has many fine features, and many of their new ones are quite interesting.

However, they have made this business of SIP URI calling too complicated.

Take a look at CallCentric for making these SIP URI calls to your friends and family.

With CallCentric, you can sign up for their free "IP Freedom" plan, where inbound and outbound SIP URI calls are free and easy. Also easy access to SipBroker.

See:

»www.callcentric.com/faq/4/150

»www.callcentric.com/faq/4/130

In fact, It's quite simple to use SIP URIs with VoIP.ms.

Just order a Virtual SIP Number and configure it the way you want. The only restriction is that it can not be called via PSTN.

The other type of SIP URI we have is for advanced users. It's direct calls to subaccounts, but you have no control on call flow this way.

Furthermore, we all due respect, I do not believe the URLs you provided will help the user with his questions on SIP URI configuration.

Anyone having questions or issues using one of our features, feel free to send me a private message will be my pleasure to help and give instructions on our more advanced options.

Videotron: Asinine name for an Arrogant company

Sun, 17 May 2009 18:23:26 EDT

PX Eliezer : You should also post about these bums in the Canadian forums. See for example:

»Canadian Broadband

»Videotron

»Switch away from Videotron

Spread lots of words about these video turds.

-------------------------------------

And you really should file a complaint with the CRTC. At least to help make them aware of the issue:

»www.crtc.gc.ca/RapidsCCM/Register.asp?lang=E

Re: [Other] VOIP.MS

Sun, 17 May 2009 17:01:55 EDT

kaila :

said by WildChild :

Unfortunately, I'm pretty sure they have the right to define the rules this way.....

Dump them now. Keeping the number isn't worth the pain you are going through. And remind me to send you a Google Voice invite when it comes to Canada.
--
Jeff Howe
Jeff's Blog - »www.jeffhowe.net/Jeffhowe.net/Blog/Blog.html

Re: [Other] VOIP.MS

Sun, 17 May 2009 15:40:08 EDT

Mango :

said by WildChild :

Before this date, it's a breach of contract, and after the renewal, it's a breach of contract. And a breach of contract and I get hit with a $120 CAD + taxes fine...

That's insane. And I thought Telus was bad.

WildChild, have you written a review here about Videotron yet?

m.

Re: [Other] VOIP.MS

Sun, 17 May 2009 15:23:44 EDT

WildChild : Unfortunately, I'm pretty sure they have the right to define the rules this way... Telcos suck in Canada because there are too few companies and they all have a monopoly, and the CRTC is too soft with all of them...

Re: [Other] VOIP.MS

Sun, 17 May 2009 15:12:48 EDT

josephf :

said by WildChild :

Videotron only offers 1 year contracts on telephony. When I called them (many times) they always told me that I had to tell the new provider to port on may 16th, the date of the end of my contract. This way, the contract isn't renewed and there is no breach of contract. Before this date, it's a breach of contract, and after the renewal, it's a breach of contract. And a breach of contract and I get hit with a $120 CAD + taxes fine...

That's highway robbery. You must port on one specific day to avoid a penalty, not a day before nor a day after?

That's purely a scam, IMHO. It practically impossible to time a port to occur on a specific day.

Re: [Other] VOIP.MS

Sun, 17 May 2009 14:40:25 EDT

WildChild : Videotron only offers 1 year contracts on telephony. When I called them (many times) they always told me that I had to tell the new provider to port on may 16th, the date of the end of my contract. This way, the contract isn't renewed and there is no breach of contract. Before this date, it's a breach of contract, and after the renewal, it's a breach of contract. And a breach of contract and I get hit with a $120 CAD + taxes fine...

Re: [Other] VOIP.MS

Sun, 17 May 2009 13:58:47 EDT

kaila :

said by WildChild :

I initiated the LNP on the 1st week of March... I contacted voip.ms many times to tell the port had to be done on May 16th.....

Wow, yet another LNP nightmare..... I wonder if you can talk to videotron and see if they can help you in some way. Not with the port process, which is probably out of their hands, but by not locking you up for another year and possibly forwarding your calls to a voip.ms number that you could dump when the port finally completes.
--
Jeff Howe
Jeff's Blog - »www.jeffhowe.net/Jeffhowe.net/Blog/Blog.html

Re: [Other] VOIP.MS

Sun, 17 May 2009 13:42:48 EDT

sream : Any indication from videotron as to wy they have held it up for so long? 10-11 weeks seems more than ridiculous.

Re: [Other] VOIP.MS

Sun, 17 May 2009 13:39:58 EDT

WildChild : I initiated the LNP on the 1st week of March... I contacted voip.ms many times to tell the port had to be done on May 16th... Yes, this is a Canadian number.

Re: [Other] VOIP.MS

Sun, 17 May 2009 11:59:48 EDT

sapstar : My port was ordered on april 2nd (I was with Bell)

Still nothing.

I contact voip.ms support on a weekly basis. Same answer everytime. It's coming, but out of their hands.

Zzzzzzz....

Re: LNP failure...

Sun, 17 May 2009 11:47:13 EDT

PX Eliezer : I hope that your port comes through. We've always read of how porting is even a more difficult process in Canada.

It may be the old carrier that is causing the problem.

What does the new carrier say?

I hope it works out, but you still may be able to use the ATA with another provider if need be....

Re: LNP failure...

Sun, 17 May 2009 10:56:57 EDT

usa2k : So when did you start the LNP?
Not really familiar with Canadian LNP process.
Was it a Canadian number?

(OT Thought: I am amazed that everything ends up in this one thread. Things interesting, and things of no interest to me, but because I have posted previously, the site tells me of Evey post. Nobody can be OT, there have been so many subjects :))

--

Jim, VoIP since 12/2002, VOIPo 2/7/2007
FAH-Tool ... Whales ... USA2K site ... VOIPo Review

Re: LNP failure...

Sun, 17 May 2009 00:38:03 EDT

Mango : Sorry to hear that :(

I forget when you ordered your port. Did you make VoIP.ms aware of the contract date? And have you requested a refund from them?

Good luck,
m.

LNP failure...

Sun, 17 May 2009 00:14:07 EDT

WildChild : Based on the contracts I have with my current provider (Videotron) my LNP was supposed to be done yesterday so I don't get hit with a $120 fine for contract breakage (1 year contract, renewed automatically at the end unless LNP occurs)... and it wasn't. I'm not expecting anything good to happen now and I'm not really happy with this... :( $20 USD wasted to try to port my number + another $20 to try voip.ms + another $60 for an ATA that won't be useful to me...

Re: Playing with SIP URI's

Sat, 16 May 2009 15:25:32 EDT

sream : ahh. Its a quarter. Anyway I'm now using pbxes along with the incoming voip.ms uri to ring group all of my free overseas did's. Allowed me to ditch the * box so now I'm just using an old version of 3cx on an always running pc as a basic local pbx/vm.

Re: Playing with SIP URI's

Sat, 16 May 2009 15:17:53 EDT

Mango : That's another way of doing it. The way apn was trying doesn't have a monthly fee though ;)

Re: Playing with SIP URI's

Sat, 16 May 2009 15:11:37 EDT

sream :

said by apn :

I've been mucking around with SIP URI's on my PAP2T.

I understand the SIP URI format for my main account to be 123456@sip.[server_ID].voip.ms where 123456 represents my 6-digit account number and [server_ID] represents the server my main account is using e.g. ca1, us4 etc.

OK, so I setup my PAP2T for IP dialing and entered what I think is the correct SIP URI into speed dial 2 on the User2 page. However, when I punch 2# into my phone (on L2) to dial L1 (DID) via the SIP URI, I get nothing but fast-busy.

Is this a limitation of the service, or am I doing something wrong?

BTW, I've confirmed that entering my L1 DID number into the speed dial allows the call to proceed.

If the SIP URI concept (from voip.ms implementation perspective) is limited to incoming calls, then I'd like test again with a softphone from my PC.

no

»https://www.voip.ms/m/virtualnumbers.php

Re: Playing with SIP URI's

Sat, 16 May 2009 14:11:51 EDT

Mango : This is strange.

I can dial my VoIP.ms SIP URI with Zoiper, but not with my PAP2T.

However, I CAN dial my internal SIP URI (username@192.168.1.9) with my PAP2T.

said by The PAP2T log :

SIP/2.0 603 Declined<013><010>Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-d0e887e7;received=[my public IP]<013><010>From: Mango ;tag=bfed36b4d33154bfo0<013><010>To: ;tag=as2d7a9c3d<013><010>Call-ID: 9023198-d463f72b@192.168.1.10<013><010>CSeq: 102 INVITE<013><010>User-Agent: VoIPMS/SERAST<013><010>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<013><010>Supported: replaces<013><010>Content-Length: 0

My test with Zoiper was without an account configured. Could it be that the PAP2T is attempting to route the call through the VoIP.ms server instead of directly to the SIP URI? If so, how do we fix that?

m.

Re: Playing with SIP URI's

Sat, 16 May 2009 10:24:24 EDT

PX Eliezer : Voip.MS has many fine features, and many of their new ones are quite interesting.

However, they have made this business of SIP URI calling too complicated.

Take a look at CallCentric for making these SIP URI calls to your friends and family.

With CallCentric, you can sign up for their free "IP Freedom" plan, where inbound and outbound SIP URI calls are free and easy. Also easy access to SipBroker.

See:

»www.callcentric.com/faq/4/150

»www.callcentric.com/faq/4/130

Re: Playing with SIP URI's

Sat, 16 May 2009 09:26:18 EDT

apn : Placing the sip uri in the dial-plan yields the same result (fast busy).

The reason I'm testing this is that I'm looking to call frends/family (also on voip) via their SIP URI's, since these calls are not routed via the PSTN and carry much reduced billing rates ;)

Re: Playing with SIP URI's

Sat, 16 May 2009 09:11:33 EDT

burgerwars : Reply to apn:
Part of what I trimmed from my dial plan (it's using Future Nine, but the same principle should apply):

Dialing 007# will dial 123456 SIP number on Future Nine (replace with whatever number you want as a speed dial)

Dialing 030 then the number then # will dial that particular Future Nine number.

This works when dialing from my BYOD Broadvoice account with a Sipura 2100. YMMV

Re: Playing with SIP URI's

Sat, 16 May 2009 08:09:53 EDT

apn : I've been mucking around with SIP URI's on my PAP2T.

I understand the SIP URI format for my main account to be 123456@sip.[server_ID].voip.ms where 123456 represents my 6-digit account number and [server_ID] represents the server my main account is using e.g. ca1, us4 etc.

OK, so I setup my PAP2T for IP dialing and entered what I think is the correct SIP URI into speed dial 2 on the User2 page. However, when I punch 2# into my phone (on L2) to dial L1 (DID) via the SIP URI, I get nothing but fast-busy.

Is this a limitation of the service, or am I doing something wrong?

BTW, I've confirmed that entering my L1 DID number into the speed dial allows the call to proceed.

If the SIP URI concept (from voip.ms implementation perspective) is limited to incoming calls, then I'd like test again with a softphone from my PC.

Re: [Other] VOIP.MS

Tue, 12 May 2009 00:57:49 EDT

sream :

said by abcezas123 :

Anyone unable to receive inbound calls tonight? I'm using IAX2 on the toronto server. My failover to a cellphone is not working either.

Been making/receiving calls all night through sip on us4 (New York).

Re: [Other] VOIP.MS

Tue, 12 May 2009 00:48:51 EDT

abcezas123 : Anyone unable to receive inbound calls tonight? I'm using IAX2 on the toronto server. My failover to a cellphone is not working either.

Re: [Other] VOIP.MS

Mon, 11 May 2009 20:05:29 EDT

PX Eliezer : Thanks for letting us know.

Much appreciated.

Re: [Other] VOIP.MS

Mon, 11 May 2009 14:42:10 EDT

papaskitch : Well, anyone hoping for CID name display on outgoing calls can forget about it for now.

Dear Customer,

Unfortunately, the carrier that used to provide CNAM on outbound calls do not provide the service right now. We tried to look for one but it happens to be unavailable at the moment. Thank you for your patience.

Sincerely,

Vincent Neupane

Re: [Other] VOIP.MS

Sun, 10 May 2009 12:30:38 EDT

hockeynomad : Wonder when Call Conferencing will be available

Quote of the day from Steve

Fri, 08 May 2009 14:43:14 EDT

Mango : I was just talking to Steve in Live Chat and found the following wildly funny. Looks like Steve has the same opinion of IE as I do. Yay for techs with a sense of humor :D

Steve Poirier: We'll also investigate the Firefox issue with the file type and fix it sometime today
Mango: Ok, no rush on that one since I can just use IE.
Steve Poirier: We don't want to torture you too much with IE!
Mango: ROFL
Steve Poirier: So we'll take a look quickly! :)

In other news: on May 20 @ 20:00 PM EST, the IP address for the Los Angeles (US3) server will be changing. If you connect by IP and wish to switch to the new server now, the new server is already operational and is named US3B. If you are using the hostname (sip.us3.voip.ms or iax2.us3.voip.ms), you do not need to do anything.

m.
--
Mango's recommended PAP2T settings: »www.toao.net/25/linksys-pap2t-vo···-review/
Linksys/Cisco dial plan tips and tricks: »www.toao.net/108/cisco-dial-plan···-tricks/

Re: [Other] VOIP.MS

Fri, 08 May 2009 13:10:20 EDT

sream : Just got an email that voip.ms changed a lot of its international rates. Very nice for me London dropped from .0138 to .007 . Looks like some overseas mobile went up. Here's the full list of changed premium rates

»www.voip.ms/rates_premium_2009-0···nges.txt

Re: [Other] Selective Call Blocking (Blacklisting)

Thu, 07 May 2009 20:13:00 EDT

N9MD :

said by Mango :

The feature was released on the 28th - Vince was entirely correct :)

said by MartinM :

The feature was published April 28th. We didn't send a newsletter yet but it's in the news when you login to the portal. That's why they informed you that the feature was not yet available.

Man, you guys are good! I believe April 28 was my day to take an extended nap ... so I missed the day's events. Thanks for the update!! :)

Re: [Other] VOIP.MS

Thu, 07 May 2009 16:01:14 EDT

hockeynomad : :)
OK, some callerid display as 01-555-555-1212.

A call filter set at 555-555-1212 will block the 01 (overseas?)

Re: [Other] VOIP.MS

Thu, 07 May 2009 12:43:22 EDT

vonnaone :

said by hockeynomad :

Yes I have noticed the same characteristic; phone displays "blocked" but the number and name shows in CDR.

Just found out from support and Martin that:

"It is probably your adapter that is configured to hide the number when the name shows as anonymous. We do pass the callerid # properly.
I believe the firmware in the pap2 is programmed to behave that way."

Oh well, it would have been nice to see those anonymous callers though.

Re: [Other] VOIP.MS

Thu, 07 May 2009 12:37:56 EDT

Mango : Unfortunately, a VoIP provider typically requires a CLEC to have presence in the appropriate rate centre, which is why no VoIP provider will port from a rate centre that only has Bell.

According to »Re: Portability Issue, rural phone companies in the US have special regulations that apply to them to prevent competition. Does anyone here know how this works in Canada?

Re: [Other] VOIP.MS

Thu, 07 May 2009 12:10:09 EDT

anon :

said by sapstar :

said by anon9 :

Excellent news, Martin, but any chance we'll see similar benefits on CDN DID's any time in the near future?

I'm curious on that one as well, Martin.

Thanks for being here for us!

+1
It would also be great if you could offer DIDs in as many Canadian rate centers as possible, and even those that aren't possible yet :) Then you could become the provider with most Canadian DID coverage and make folks in un(der)served areas (like me) very happy! There are some rate centers that only have Bell/Rogers currently, and no VoIP provider will port from those exchanges. It's really a bummer.

Re: [Other] VOIP.MS

Thu, 07 May 2009 11:56:39 EDT

MartinM : I thought I would take some advance and post this before someone ask about it:

The Planet is currently experiencing issues with one of their data centers. This affec tone of our rack at the planet, specifically: Website and main Database.

Emails and VoiP servers are not affected by this. Each VoIP servers is configured to work with its own local copy of the database.

~~Website is down~~ but VoIP communications are NOT affected.

Thank you for your patience.

Edit: It's already back up..

Re: [Other] VOIP.MS

Thu, 07 May 2009 10:21:42 EDT

MartinM : We don't have enough volume yet on Canadian DIDs to offer price reductions yet but we're getting there.

We do have something that we're trying to push for this week for residents of Quebec: Around ~200 new rate centers.

New incoming coverage:

chomedey
st-jerome
albanel
alma
anse-st-jean
arthabaska
aylmer
baie-st-paul
beauharnois
beloeil
boischatel
boucherville
bromont
bromptonville
brownsburg
buckingham
cabano
cap-st-ignace
chambly
chambord
charny
chateau-richer
chateauguay
chelsea
chesterville
chicoutimi
clermont
coaticook
contrecoeur
coteau-du-lac
coteau-landing
cowansville
crabtree
deauville
delisle
deschaillons
dolbeau
donnacona
drummondville
dunham
east angus
farnham
gatineau
granby
howick
hudson
hull
ile-aux-coudres
ile-perrot
joliette
jonquiere
l’epiphanies-l ou assomption
la baie
la guadeloupe
la malbaie
la pocatiere
lac-megantic
lachine
laprairie
laterriere
laval-est
laval-ouest
lavaltrie
le gardeur
les cedres
les eboulements
levis
longueuil
loretteville
luskville
magog
maniwaki
marieville
mascouche
milot
mirabel-aeroport
mirabel-st-augustin
mirabel-ste-scholastique
mont-laurier
montebello
montmagny
montrÃ©al
neuville
norbertville
normandin
north hatley
notre-dame-de-stanbridge
notre-dame-des-laurentides
notre-dame-du-lac
oka
papineauville
peribonka
perkins
pointe-claire
pont-viau
portneuf
princeville
rawdon
rigaud
riviere-du-loup
roberval
rock island
roxboro
shawbridge
sherbrooke
sorel
st-agapit
st-alexandre
st-alphonse-de-rodriguez
st-ambroise
st-andre
st-andre est
st-andre-avellin
st-anselme
st-antoine-de-tilly
st-apollinaire
st-augustin-de-desmaures
st-basile-de-portneuf
st-blaise
st-bruno
st-calixte-de-kilkenny
st-cesaire
st-charles-de-bellechasse
st-clet
st-constant
st-damien
st-denis
st-edouard-de-lotbiniere
st-eustache
st-felicien
st-felix-de-valois
st-fereol
st-fidele
st-flavien
st-francois
st-fulgence
st-henri-de-levis
st-hilarion
st-honore chicoutimi
st-honore-de-temiscouata
st-hyacinthe
st-jaques
st-jean
st-jean ile-d’orleans
st-jean-port-jolie
st-lambert
st-lambert-de-lauzon
st-lin
st-marc
st-michel-de-bellechasse
st-nicolas
st-ours
st-pacome
st-pascal
st-paul-d’abbotsford
st-philippe-de-neri
st-pie
st-polycarpe
st-prime
st-raphael-de-bellechasse
st-raymond
st-remi
st-roch-de-aulnaies
st-sauveur
st-simeon
st-tite
st-tite-des-caps
st-vincent-de-paul
ste-anne-de-beaupre
ste-anne-des-plaines
ste-catherine
ste-claire-de-dorchester
ste-croix
ste-genevieve
ste-julie-de-vercheres
ste-julienne
ste-martine
ste-petronille
ste-rose
ste-therese
ste-victoire
stoke
stoneham
tadoussac
terrebonne
thedford mines
thurso
trois-rivieres
val-alain
valcartier
valleyfield
varennes
vaudreuil
vercheres
victoriaville
ville-degelis
waterloo
waterville
windsor
yamaska

We'll keep you updated for other provinces. We're always working on aquiring more coverage.

Re: [Other] VOIP.MS

Thu, 07 May 2009 09:56:47 EDT

sapstar :

said by anon9 :

Excellent news, Martin, but any chance we'll see similar benefits on CDN DID's any time in the near future?

I'm curious on that one as well, Martin.

Thanks for being here for us!

Re: [Other] VOIP.MS

Thu, 07 May 2009 09:47:42 EDT

Boolah : PM'ed ticket numbers. Thanks Martin!

Re: [Other] VOIP.MS

Thu, 07 May 2009 09:33:30 EDT

MartinM : What are your ticket numbers boolah?

Re: [Other] Selective Call Blocking (Blacklisting)

Thu, 07 May 2009 09:29:41 EDT

MartinM :

said by N9MD :

You might want to
update your support staff. ;)

The feature was published April 28th. We didn't send a newsletter yet but it's in the news when you login to the portal. That's why they informed you that the feature was not yet available.

Re: [Other] VOIP.MS

Thu, 07 May 2009 08:39:58 EDT

burgerwars :

said by kaila :

Just received an loooong email from voip.ms announcing lower DID rates, then listing what must be their entire effected customer bases DID's- oops! Most DID's (USA) including mine, seem to have been lowered to .99mo with a 1 cent per minute incoming rate, or $4.95mo on the flat plan. Some of the hot rate centers (Miami, DC, San Francisco, etc) remain far higher.

I love 'em, but their margins have to be razor thin.

I got the same long e-mail plus two small ones. I was wondering why the long e-mail I couldn't open. I get it in Yahoo mail and thought the size it said was 21KB. Then I looked again and noticed it was actually 21MB. That explains it.

Re: [Other] VOIP.MS

Thu, 07 May 2009 07:58:50 EDT

Boolah : Martin - Do you have a time frame of when your web-based ticketing system (mentioned in this post) will be available? I've got three tickets going to back to March 10th that haven't been responded to at all (other than the automated "we have received your inquiry" email).

When going to live chat and referencing the ticket numbers, I'm told that they will look into the issues and get back to me by the end of the day. But I still haven't received a response. The last time I was told this in live chat was April 1st (perhaps it was just an April fool's joke :))

Re: [Other] Selective Call Blocking (Blacklisting)

Thu, 07 May 2009 01:16:27 EDT

Mango : The feature was released on the 28th - Vince was entirely correct :)

m.

Re: [Other] Selective Call Blocking (Blacklisting)

Thu, 07 May 2009 00:52:03 EDT

N9MD : Thanks for the prompt reply, Martin. Here's my April 22 ticket submission followed by the response I got
on April 23 from your rep:


My Question:
.
Hello people.
.
My Voip.ms DID is 678-xxx-xxxx ... SIP User Name ######.
.
I am constantly receiving nuisance calls from 866-675-1540,
a Call Center apparently repre senting Financial Management
Services which is shown to be a scamming company in my
Google  search for this name and number.
.
I've searched the Voip.ms webpages but have not been able
to find an option for blacklisti ng specific numbers.  Do
you provide this option?
.
N9MD
.
.
.
Voip.ms Response:
. 
REFERENCE NUMBER: LTK151031171867X 
. 
Dear N9MD, 
. 
We will have a call blocking/filtering feature in the near
future but right now we do not  have this available.
. 
Sincerely, 
Vincent Neupane
VoIP.ms Customer Service

.
Your response in this thread did indeed lead me to the
Blacklisting area. Thanks!

You might want to
update your support staff. ;)

Re: [Other] Selective Call Blocking (Blacklisting)

Wed, 06 May 2009 23:53:17 EDT

MartinM :

said by N9MD :

Martin ... while we're discussing your plans for new features ... please give some deep though to providing Selective Call Blocking (Blacklisting).

My Voip.ms DID# was issued to me last December from a new number block assigned to XO by NANPA. So, most likely, my DID was not a previously used number.

From early February onward, I've been getting calls constantly from 866-675-1540 ... a sleazy Oklahoma collection agency (according to my on-line sources for this number). Since I have my Voip.ms DID permanently forwarded to my office PSTN, I actually have never had the pleasure of talking with these lowlifes. They either get a member of my office staff or my answering service, all of whom just hang up on the caller.

My point --- With my other providers, I can put the offending DID into the Blacklist, giving them a busy signal for their troubles. How soon will Voip.ms have this feature in place?

866-675-1540 678-298-7000

It's already in the customer portal under DID Management -> CallerID Filtering

Re: [Other] Selective Call Blocking (Blacklisting)

Wed, 06 May 2009 23:00:20 EDT

N9MD : Martin ... while we're discussing your plans for new features ... please give some deep though to providing Selective Call Blocking (Blacklisting).

My Voip.ms DID# was issued to me last December from a new number block assigned to XO by NANPA. So, most likely, my DID was not a previously used number.

From early February onward, I've been getting calls constantly from 866-675-1540 ... a sleazy Oklahoma collection agency (according to my on-line sources for this number). Since I have my Voip.ms DID permanently forwarded to my office PSTN, I actually have never had the pleasure of talking with these lowlifes. They either get a member of my office staff or my answering service, all of whom just hang up on the caller.

My point --- With my other providers, I can put the offending DID into the Blacklist, giving them a busy signal for their troubles. How soon will Voip.ms have this feature in place?

866-675-1540 678-298-7000

Re: [Other] VOIP.MS

Wed, 06 May 2009 22:08:31 EDT

anon : Excellent news, Martin, but any chance we'll see similar benefits on CDN DID's any time in the near future?

Re: [Other] VOIP.MS

Wed, 06 May 2009 20:12:19 EDT

MartinM : We apologize for the emailing glitch :) We tested with a single user and the email system worked properly but a glitch was introduced when we created the loop and many bogus dids were included in our first copy including numbers in stock not even assigned.

We've resent the proper version. We apologize for the confusion caused by the bugged out email.

Regarding new rates, we've reached high volume with some carriers and vendors and instead of pocketing the extra money on our new purchasing power we simply decided to give it back to the customer base by lowering the rate centers we could, including the DIDs that were already sold.

Regarding hot centers, some of them were changed. For example, if you look at our list of available DIDs at the moment for Miami, some are listed at 0.99 and some at 1.49. This mean that some of those DIDs are not eligible, it can be for various reason but the main one is that's it's on a different carrier. So the change is not only rate center specific, it's also DID specific.

Let me know if you have any more questions regarding this change.

Re: [Other] VOIP.MS

Wed, 06 May 2009 19:53:14 EDT

kaila : Just received an loooong email from voip.ms announcing lower DID rates, then listing what must be their entire effected customer bases DID's- oops! Most DID's (USA) including mine, seem to have been lowered to .99mo with a 1 cent per minute incoming rate, or $4.95mo on the flat plan. Some of the hot rate centers (Miami, DC, San Francisco, etc) remain far higher.

I love 'em, but their margins have to be razor thin.
--
Jeff Howe
Jeff's Blog - »www.jeffhowe.net/Jeffhowe.net/Blog/Blog.html

Re: [Other] VOIP.MS

Tue, 05 May 2009 17:08:47 EDT

anon : Hmmm, then it might be the firmware implementation in your phones;

IF {PRIVATE} FLAG received

display callerID as "Private"

ELSE

display callerID name & number

ENDIF

Re: [Other] VOIP.MS

Tue, 05 May 2009 14:21:10 EDT

Mango : Ah, I see. If CID is passed correctly for normal calls, then what I suggested may not help. I'll try and test this later on and see what I can come up with.

Re: [Other] VOIP.MS

Tue, 05 May 2009 12:55:41 EDT

hockeynomad : Yes I have noticed the same characteristic; phone displays "blocked" but the number and name shows in CDR.

Re: [Other] VOIP.MS

Tue, 05 May 2009 12:06:24 EDT

vonnaone :

said by Mango :

Are you saying that the number appears in CDR but does not get passed to your phones?

Thanks Mango, I'll give it a try tonight.

edit: CID is passed correctly for everything except anonymous calls which show as 'private caller'. I can see in the CDR the anonymous callers phone numbers, but those are not being passed to me.

Re: [Other] VOIP.MS

Tue, 05 May 2009 11:21:25 EDT

Mango : Are you saying that the number appears in CDR but does not get passed to your phones? If that's the case, then try the following:

Set (Regional tab) the Ring1 Cadence to 60(2/4), the Ring Waveform to Sinusoid, and the Ring Frequency to 20. Next, set the Caller ID method to Bellcore(N.Amer,China) and the Caller ID FSK Standard to bell 202.

If that doesn't work try setting the Ring Waveform to trapezoid.

Does Caller ID appear properly on the Info tab of the device if you refresh it while you are on a call?

m.

Re: [Other] VOIP.MS

Tue, 05 May 2009 10:31:03 EDT

anon : I believe Caller name and the Private flag are mutually exclusive, hence suspect that your ATA/phone are accurately displaying what's given.

It might be that they are adding 'Anonymous' to the CDR when they have no name info to display.

Re: [Other] VOIP.MS

Tue, 05 May 2009 08:50:42 EDT

vonnaone : Has anyone noticed that your CID on the phone displays 'Private Caller' but when you check the CDR reports online it gives you the phone number and 'Anonymous'?

I'm using a SPA9102. I wonder why CID isn't being passed to my device....Anyone know?

Re: [Other] VOIP.MS

Sat, 02 May 2009 08:52:10 EDT

papaskitch : I've confirmed with support, that some lucky few are still using the old carrier (not sure of the exact reason) which supported sending of CID name. They're investigating getting me on that route as well. Although in my case, if they get me on that route, I would be on the hook for additional per-minute costs.

Re: [Other] VOIP.MS

Sat, 02 May 2009 08:49:07 EDT

papaskitch : With the PAP2T, under the User1 tab, there's a 'Default Ring' setting. I would guess it's very similar with the 2102. On the same tab, there's a 'Distinctive Ring' section where you can specify which numbers get which rings (including a mother-in-law).

Re: [Other] VOIP.MS

Sat, 02 May 2009 07:49:22 EDT

hockeynomad : My Linksys 2102 came preset with all 8 Ring Cadence with different settings. How to know which is being used?

Ring Cadence 1 is set at 60(2/4)
Ring Cadence 2 at 60(.8/.4,.8/4) and so on

I think if your really get into the head of this adapter you can also block a mother-in-law

Re: [Other] VOIP.MS

Fri, 01 May 2009 21:43:26 EDT

Mango :

said by papaskitch :

From voip.ms support with regards to outgoing CID name:

We're trying to find a vendor that can matches the price we're selling. For now we don't support it.

Bummer.

I was curious about this so I did some tests yesterday. I hooked up a regular phone to a line from Shaw Cable, our CLEC. I used the Premium and Value routes on the CA1, CA2, and US3 servers. In all tests except one, the name was sent along with the number. Once on US3/Premium it came through as "Brit. Columbia" but I was not able to duplicate this with further calls.

Some time back I was switched to their "second main carrier" due to a routing issue. Perhaps this is the reason why it's working for me. But if this were the case, I'm not sure why they would be still trying to find a vendor.

m.

Re: [Other] VOIP.MS

Thu, 30 Apr 2009 09:02:15 EDT

papaskitch :

said by Mango :

I just thought of one other thing that may help:

said by www.toao.net :

The PAP2T we received shipped without a North American ring. We were able to achieve a "normal-sounding" ring by setting (Regional tab) the Ring1 Cadence to 60(2/4), the Ring Waveform to Sinusoid, and the Ring Frequency to 20. We're told that these settings are standard Bellcore settings, though we've also had reports of specific telephones that respond better to a Trapezoid Ring Waveform. Try Sinuoid first, and if your phone doesn't ring properly or you have Caller ID issues, try Trapezoid instead.

I remembered to try this last night, and switching the Ring Waveform to Sinusoid did the trick. Thanks!

More new features...

Wed, 29 Apr 2009 11:56:40 EDT

Mango :

Regarding Voicemail, when Dialing *97, if your account or subaccount has a mailbox associated to it, you will be prompted directly for the password and you will not have to enter the mailbox id. If you wish to access a different mailbox than the one associated to your account or sub account, dial *98.

I have verified that this is working :D

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 16:36:33 EDT

Mango : I would imagine it means "any calls that do not come with Caller ID number".

Rumor has it that they are working on feature codes. :)

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 16:10:28 EDT

hockeynomad : CallerID filtering; this must be indicated in your online profile. My previous provider had *60 and you followed the prompts.

Unfortunately, that was the only redeeming feature of that provider.
------------------------------------------------------------
"Anonymous call blocking" not sure exactly what that means.

1. Blocked calls
2. Private name

I sure this is not a blanket catch all for telemarketers.

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 15:52:12 EDT

Mango : I have successfully used Toronto, Montreal, and Los Angeles. Sorry I can't offer any more advice; I'm as confused as you are!

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 15:47:34 EDT

papaskitch :

said by Mango :

I used my PAP2T ATA to set the Caller ID Name, and the VoIP.ms control panel to set Caller ID Number. One note: outgoing CID Name does not work reliably on the Value routes, only on Premium routes. I am not sure why outgoing CID Name should not work for you as I assume you and I, both being Canadian, would use the same carrier. I am open to corrections on this however.

m.

Strange, I have the same setup (premium route included). Moreover, it shouldn't work according to voip.ms support staff.

I'm currently using the Montreal server although I've also tried the Toronto server with similar results. Which server are you using? Perhaps MartinM can chime in on why this might be the case for some, and not for others.

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 15:30:26 EDT

Mango :

said by papaskitch :

Now, if I could add a CallerID filter directly from the CDR, that would be perfect.

Eh, there's always copy-and-paste. :D I'm just happy to have the feature available - this is a great feature that for some odd reason many phone companies just don't have.

said by papaskitch :

BTW, Mango2, you mentioned that outgoing CID name works for you. Is this still the case, as mine definitely doesn't work (and shouldn't work according to voip.ms support). Are you using an ATA, or do you have an Asterisk setup?

I used my PAP2T ATA to set the Caller ID Name, and the VoIP.ms control panel to set Caller ID Number. One note: outgoing CID Name does not work reliably on the Value routes, only on Premium routes. I am not sure why outgoing CID Name should not work for you as I assume you and I, both being Canadian, would use the same carrier. I am open to corrections on this however.

m.

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 15:05:10 EDT

papaskitch : Now, if I could add a CallerID filter directly from the CDR, that would be perfect. For that matter, being able to listen to a voicemail via the CDR would be really nice too.

BTW, Mango2, you mentioned that outgoing CID name works for you. Is this still the case, as mine definitely doesn't work (and shouldn't work according to voip.ms support). Are you using an ATA, or do you have an Asterisk setup?

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 14:30:21 EDT

Mango :

New Feature: CallerID Filtering and some changes to Time Conditions

**SQUEAL~!!1**

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 11:23:49 EDT

Mango : I just thought of one other thing that may help:

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 11:05:26 EDT

papaskitch : Thanks for the tip. I'll definitely play with that tab when I get home.

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 10:43:32 EDT

Mango : Interdigit Short Timer is for dialing, not incoming calls.

Distinctive rings may be configured on the User tab. For example, my area code is 604, so if I place 604* in Ring2 Caller and anyone calls from my area code, the phone will play Ring2.

Note that the distinctive ring patterns may be configured on the Regional tab.

Hope that helps!

m.

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 10:37:53 EDT

papaskitch :

said by Mango :

How does a long distance call ring?

Two quick pulses as opposed to one long one. I've read briefly that the Interdigit Short Timer should be set to 5 (it's currently at 3). Perhaps that will help.

After doing a little more reading, that doesn't look like it will help at all.

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 10:34:42 EDT

Mango : How does a long distance call ring?

Re: [Other] VOIP.MS

Tue, 28 Apr 2009 10:30:18 EDT

papaskitch : Can anyone tell me why my incoming calls ring like they're long distance? I'm using the PAP2T and am guessing it might having something to do with the way it's setup.

Re: [Other] VOIP.MS

Fri, 24 Apr 2009 20:40:55 EDT

Mango :

said by MartinM :

Regarding the CNAM change for Canada (No queries when Name is received), we're investigating and if there's any problem, it should be corrected within 1-2 days.

This is now working :D YIPPEE!!!! I'll update my review shortly.

Re: [Other] VOIP.MS

Fri, 24 Apr 2009 08:20:15 EDT

papaskitch : Just to clarify, incoming Caller ID name works. It's the outgoing name that is a problem.

It hasn't been working since I signed up. In fact, I was told that outgoing caller ID name is known not to work (maybe because I'm using a Canadian DID?). The number gets sent, but it's always accompanied by 'unknown name'. Enabling CallerID Name Lookup won't help with outgoing calls will it?

From voip.ms support with regards to outgoing CID name:

We're trying to find a vendor that can matches the price we're selling. For now we don't support it.

Bummer.

Re: [Other] VOIP.MS

Fri, 24 Apr 2009 03:58:13 EDT

stoked : My number just got ported on Wednesday, so far so good. However, it looks like the voip.ms DNS servers are down as of 12:57AM Friday April 24th.

Re: [Other] VOIP.MS

Fri, 24 Apr 2009 01:02:11 EDT

Mango :

said by MartinM :

Regarding the CNAM change for Canada (No queries when Name is received), we're investigating and if there's any problem, it should be corrected within 1-2 days.

Good to hear :) (FYI, if it helps, I tested it with a VoIP.ms subaccount and also a phone line from Shaw Cable.) And I think I've said this before, but it's good to see you here in the forums Martin - I should think that monitoring forums would go a long way in becoming one of the top rated PAYG/BOYD customer service in the industry. :D

It's possible my wife may be getting a job in a city about 100km from here. Being able to keep our current phone numbers so our friends can call us toll free will be quite the concept! I just love this technology.

Has anyone set up IVR yet?

m.

Re: [Other] VOIP.MS

Thu, 23 Apr 2009 23:12:37 EDT

MartinM : Regarding the CNAM change for Canada (No queries when Name is received), we're investigating and if there's any problem, it should be corrected within 1-2 days.

CallerID filtering is in testing, and it's gonna be a very powerful feature. My guess is 1-2 days. Should be online tomorrow.

We also hired one programmer that is dedicated to building a new in-house web ticketing system for support. There's no need to "witch hunt". We're aware that we've outgrown the capabilities of our emailing system and that some escalated issues are taking longer than what we want to offer to our customer base. We're working hard on this. We're not ignoring anyone issues, we just receive hundreds of them per day , 95% not related to issues on our side. We believe our new system will allow us to offer one of the top rated PAYG/BOYD customer service in the industry we're looking only at a few weeks for this. We also extended hours on live chat, you should be able to speak with an agent up to 10PM from Monday-Fri.

We're very very busy at VoIP.ms at the moment developing and extending our offering, I don't have much time to read the forums, You can all contact me to this email: martin@voip.ms, it will be my pleasure to help you.

Re: [Other] VOIP.MS

Thu, 23 Apr 2009 22:45:25 EDT

Mango : Outgoing Caller ID Name seems to be working for me. In fact I never noticed a problem. How long has it not been working for you, and are you using Premium routing?

Re: [Other] VOIP.MS

Thu, 23 Apr 2009 16:06:12 EDT

papaskitch : Has anyone heard about any plans to get outgoing name display working again? I can understand why it stopped working, but it would be nice to know that they're working on getting it going again. I find my friends tend to ignore the 'unknown name' on occasion (or maybe they just don't like me :()?

I've sent off the same question to support, but haven't heard back (it's only been a day) and was hoping someone else has asked this question more recently.

Re: [Other] VOIP.MS

Wed, 22 Apr 2009 23:01:16 EDT

Mango :

The other change is a modification made for our Canadian DID Numbers. If the system receives a CallerID name when a call is received, we'll not Query the CNAM database even if you have the CNAM feature Activated.

I have not been able to make this work. I reported it to Stephen who decided to escalate the issue.

Fingers crossed :)

Re: [Other] VOIP.MS

Wed, 22 Apr 2009 21:38:49 EDT

anon : The official announcement:

We released 3 new features today: Digital Receptionist (IVR), Calling Queues and Recordings. We're now working on the CallerID Filtering and this feature should be online this week.

We've made 2 modifications to the way the CNAM system works. First modification is a 24 hours caching system. If you receive more than 1 call from the same CallerID Number in the same day, you will be charged only for one query. The other change is a modification made for our Canadian DID Numbers. If the system receives a CallerID name when a call is received, we'll not Query the CNAM database even if you have the CNAM feature Activated.

Re: [Other] VOIP.MS

Wed, 22 Apr 2009 19:01:21 EDT

anon : Take a look at the user portal .. new features appeared today.

IVR (Digital Receptionist)
Calling Queues
Recordings

However, no sign of Selective Call Blocking or Anonymous Call Reject .... bummer.

Re: [Other] VOIP.MS

Fri, 17 Apr 2009 13:44:42 EDT

vilord :

said by HappyBear :

But outbound usually get word chipped off.

check to be sure your udp packet size is set to .020, not .030, it seems like the voip.ms servers cannot handle .030, and they end up chopping off part of the packets resulting in choppy audio.

Re: [Other] VOIP.MS

Fri, 17 Apr 2009 09:30:00 EDT

hockeynomad : Informed chat support that I have been unable to access saved voice messages. The prompts do not lead me there unless I missed something.

They told me perhaps it was saved on another server. Maybe, as I used the Montreal server till the Toronto one appeared.

Follow-up told me that the message is indeed in the current Toronto server, so again I asked how am I able to access saved voice messages. I am still waiting.

Re: [Other] VOIP.MS

Wed, 15 Apr 2009 17:10:11 EDT

Mango : You may wish to look at »[General] Port number from defunct VoIP carrier? . Some providers have reported success porting numbers away from already defunct providers.

Re: [Other] VOIP.MS

Wed, 15 Apr 2009 14:49:06 EDT

abcezas123 : Reformatted my box with a fresh copy of piaf 1.4 and now it works, no problems.

Question regarding LNP - let's say I port my number currently with Rogers Home Phone (POTS) to voip.ms and heaven forbid, they (voip.ms) go under. What happens to my number? Will I be able to transfer it to another provider, or will it go back into a public pool?

Re: [Other] VOIP.MS

Wed, 15 Apr 2009 12:56:21 EDT

anon : Thanks mango2, I will contact them to see if anything can be done.

Re: [Other] VOIP.MS

Wed, 15 Apr 2009 12:09:34 EDT

Mango : After I reported the issue to Support enough times, they "made a change to [my] routing" and the problem vanished. Hopefully the solution for you will be just as simple.

Re: [Other] VOIP.MS

Wed, 15 Apr 2009 11:57:00 EDT

PX Eliezer :

said by HappyBear :

I've been using the VOIP.MS for 1.5 months and it seems the sound quality is getting worse recently. Inbound quality is good. But outbound usually get word chipped off.

My previous experience with them was the same.

Re: [Other] VOIP.MS

Wed, 15 Apr 2009 10:39:17 EDT

anon : I've been using the VOIP.MS for 1.5 months and it seems the sound quality is getting worse recently. Inbound quality is good. But outbound usually get word chipped off. I started to switch to premium line but the overall quality is still not as good. I uses comcast HSI so the connection speed does not look like an issue. I am debating if I would still continue the service after the $25 initial deposit used up.

Re: [Other] VOIP.MS

Tue, 14 Apr 2009 19:26:56 EDT

Mango : Problem has been resolved. Issue was due to a "minor problem on the network" according to support.

Re: [Other] VOIP.MS

Tue, 14 Apr 2009 19:14:30 EDT

Mango : I am unable to make calls right now. Calls to the echo test and Voicemail work, but not calls to the PSTN. I tried the US3 and CA2 servers. Incoming calls still work fine.

604-299-9000 is a test number if anyone else wants to try.

Re: [Other] VOIP.MS

Tue, 14 Apr 2009 15:31:17 EDT

usa2k : It just feels like it goes with the BITE comment :)

»www.youtube.com/watch?v=SXn2QVipK2o

Re: [Other] VOIP.MS

Tue, 14 Apr 2009 13:48:01 EDT

Mango : I just spoke with Peter on Live Chat, who I assume is one of the new people Martin spoke of. Welcome, Peter! We don't bite...much ;)

Re: Just got an email from MartinM...

Mon, 13 Apr 2009 20:11:21 EDT

Darkev : That's good news. I don't get a lot of calls, but even 5 cents is a good savings.

I wonder if voip.ms will ever charge in Canadian funds. That's the only thing I really find wrong with the service. Considering our dollar is currently only worth .75 cents on the US dollar, the $8.95 charge for DID comes out to be ~$11.19 per month, and there are taxes on top of that. For incoming calls only, it's a bit pricey. If it was $8.95 Canadian it would be more on par with what other Canadian companies are charging for VoIP. In fact, some charge around $15 for unlimited Canada/US outgoing as well as unlimited incoming. So the cost of voip.ms is rather high. The quality is equivalent to a couple of other providers I've tested. I have Primus too, which is a bit more expensive, but they have many features that none of the other companies offer. The disappointment for Primus is they don't offer BYOD. Once they do that, they'll reach #1 in my book.

Re: Just got an email from MartinM...

Mon, 13 Apr 2009 19:21:38 EDT

Mango : According to Martin, they are making two improvements to Caller ID. The first is that they are caching CNAM queries so that if the same person calls you multiple times in a 24 hour period, you only get charged for one lookup. The second improvement is that they are setting the system up to display passed Caller ID for Canadian DIDs, and do a CNAM query only if it does not exist. This should reduce the calls that display "CELL PHONE BC" or "GATINEAU QC".

m.

Re: Just got an email from MartinM...

Mon, 13 Apr 2009 19:09:41 EDT

Darkev :

said by Mango :

"[The new Caller ID features] will be ready this week along with other features such as callerid filtering, ivr, queue and system recordings. We'll also cache cnam entries on a 24 hours period."

Hi Mango2,

What new caller id features? I haven't heard anything about this. I currently have caller id with voip.ms, but pay for each lookup. Are they relaxing the charge on looking up the name? What do you mean when you say, "cache cnam entries"? What are the benefits of these changes to us?

Thanks.

Re: [Other] VOIP.MS

Mon, 13 Apr 2009 19:07:29 EDT

Darkev :

said by Mango :

Good to hear. How do you like your SPA9000? Do you use its PBX features?

Yes, I absolutely love my SPA9000. I'm always discovering something new with it. One of the things I didn't like about it was the inability to retrieve voicemail from companies like voip.ms, however I figured out how to do it. Now, when someone leaves me voicemail on voip.ms, the red lights illuminate on all of my SPA942 phones.

It is a fantastic device, and is very flexible. It's installed in my home, but this device would be quite beneficial in a small business as it supports up to 16 IP phones, and 2 analog phones.

Many of the settings are identical to the settings of the PAP2T. So if you are having trouble staying registered, factory reset your equipment and then add the 3 or 4 changes that you need for voip.ms and you should be good to go.

Just got an email from MartinM...

Mon, 13 Apr 2009 17:57:50 EDT

Mango : "[The new Caller ID features] will be ready this week along with other features such as callerid filtering, ivr, queue and system recordings. We'll also cache cnam entries on a 24 hours period."

I for one am looking forward to it :D

Re: [Other] VOIP.MS

Mon, 13 Apr 2009 17:35:42 EDT

Mango : Good to hear. How do you like your SPA9000? Do you use its PBX features?

Re: [Other] VOIP.MS

Mon, 13 Apr 2009 16:14:03 EDT

anon : Hello,

I was having a registration problem with voip.ms and fixed it. It had nothing to do with the registration interval. It was one of the settings on my Linksys SPA9000 device. Which setting is a good question. I was originally with another VoIP provider before switching to voip.ms. This other provider had me change several SIP options. That's why my registration was unstable with voip.ms. Unfortunately, the illustrations on the voip.ms website are not advanced settings. If your advanced settings are messed, the voip.ms illustrations won't help.

To fix my problem, I did a factory reset on my voip equipment. This restored all settings to the original Linksys settings. I then added the voip.ms proxy, user, and password. Since then, I have never gone down once.

I will also mention that I don't have NAT turned on. I left it off and have had no trouble. Not sure why I would need it on.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 22:00:47 EDT

fabrini :

said by Mango :

This can happen when your registration interval is too long. Basically, VoIP.ms thinks you're still registered because you haven't missed a registration. Then the registration expires and your box re-registers.

This makes sense, rather then monkeying with the refresh rate, I read a little about IAX2. I decided to try running my connection to VOIP.ms with IAX2 instead of SIP. I'll post if there is any improvement.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 19:30:38 EDT

Mango :

said by fabrini :

What I don't understand is why both my box, and the voip.ms admin panel were showing successful registration and I was still getting the busy.

This can happen when your registration interval is too long. Basically, VoIP.ms thinks you're still registered because you haven't missed a registration. Then the registration expires and your box re-registers. Calls made shortly after registration should go through, which is why the problem appears to happen only some of the time.

You may want to set up a voicemail account as backup and have it email you new voicemail. That way, when clients call, at least you can get a message and call them back.

m.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 13:56:00 EDT

N9MD :

said by WildChild :

My dial plan is good, the fast busy is on inbound calls, is intermittent and it only happens with calls from a particular provider.

Your comment that "it only happens with calls from a particular provider" gives a powerful clue to the underlying problem.

As reported in the past ... but not recently ... failed incoming calls coming from a particular provider are usually due to a DID not being registered with the calling party's provider's "Routing Tables". When a caller dials your number, his/her telephone provider must first look at one of many Routing Tables (nationwide or regional) that are maintained by a variety of companies. This data dictates the routing to your number (whether it be land line, cellular, beeper, VoIP).

For example, I obtained a DID from a VoIP carrier a year or two back ... from a newly released, never before assigned number block (assigned to the VoIP provider's DID supplier ... a CLEC such as Level 3 Comm... by NANPA). The routing info for this DID took a while to circulate among the routing servers ... but apparently it never got added to the Verizon Wireless database in the SouthEast (specifically Florida). So my brother could reach my NJ VoIP number from his land line but not from his Florida based Verizon cellphone. All other folks could reach me with no problems.

I subsequently ported my VoIP DID# to an ATT Mobility cellphone (in NJ) ... but my brother still could not reach me from his FL Verizon cellphone ... most likely because Verizon Wireless' routing tables were never properly updated from the get-go.

I recently suggested that my brother call Verizon Wireless, since only the calling party's provider can correct the problem. Sure enough, I received a call directly from a Verizon Wireless trouble shooter confirming my cellphone number and stating that my brother had entered a complaint with their customer service people. Within less than 5 minutes after this call, my brother called me successfully on my cellphone. Problem resolved!

So, if my reading of your plight is correct, your best bet is to have the people who cannot reach your DID complain to their phone company .... but make sure you callers do not mention the word VoIP since that tends to confuse Level 1 Customer Reps at land line phone companies. They should just state that they cannot get through to your number.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 13:54:50 EDT

fabrini : I've been using VOIP.ms for about a month as my primary incoming trunk, connecting to a FreePBX system. Virtually flawless until yesterday.

Asterisk showed successful registration on sip show registry, the voip.ms panel showed I was registered as well, but when I called my DIDs I would get a busy. According to a tech support chat session this is what happens if the phone is not registered and there is no voicemail failover. What I don't understand is why both my box, and the voip.ms admin panel were showing successful registration and I was still getting the busy. This wouldn't happen all the time, just sometimes.

The problem doesn't seem to be happening today, but I'm not getting very many calls today (most clients have good friday off) so its hard to really know.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 13:13:45 EDT

abcezas123 : Placed it into the DMZ and it craps out after ~6 minutes. :(
Another problem that I've had is that sometimes when I reload my config or restart asterisk, sometimes the voip.ms sip trunk doesn't show up under 'sip show peers', and in 'sip show registry', it shows unregistered with a reg. interval of 60, which is the default that asterisk assigns, whereas my voipbuster trunk always comes online with a reg. interval of 60, which I've set manually.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 12:57:42 EDT

abcezas123 : Hi Mango,

I'm using a sip trunk. While I've heard that IAX2 is much better with NATs, that did not work for me at all. Plus I wish voip.ms had a status page showing IAX registrations like they do with SIP. Anyway, the thing is that I have a sip trunk with another voip provider (voipbuster) and that's never lost its registration. I changed the registration interval down to 60, that did not help. I'm going to try putting the box in the DMZ and see what happens. I'll post back.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 12:44:59 EDT

Mango : Loss of registration can be a NAT issue. Are you behind a firewall? If so, have you turned on NAT in the VoIP.ms control panel? I'm afraid I'm not familiar with PIAF, but does it have NAT keep-alive settings as well?

Is it that the device tries to re-register and fails, or is it that it loses its registration well before it has a chance to re-register? If the latter, you may wish to lower your registration interval. I have success with every 300 seconds; some people only have success with every 120 seconds.

Are you using SIP or IAX? I have heard that IAX does a better job of NAT traversal. If NAT is indeed the problem, switching to IAX may help.

m.

Re: [Other] VOIP.MS

Fri, 10 Apr 2009 03:28:11 EDT

anon : I'd like to chime in about my experience with voip.ms as well. While I've found that when the service works (and that's a big when in my case), its beautiful. Crystal clear, two way audio right out of the box. But, I have had troubles with my sip trunk (i'm using PBX in a Flash running Asterisk 1.4) losing registrations very, very often. I bought a DID to test out the service before porting my residential number, and thank god I did, because so far I would have had very unreliable home phone service. I have spoken to the support team and as of yet they have no solution. Martin, my reference number for the case is LTK151030133961X. It would be appreciated if you could help me out.

Re: [Other] VOIP.MS

Wed, 08 Apr 2009 09:12:28 EDT

hockeynomad : PX, exactly. One of the main reasons I went with VOIP was the static on my end of my POTS line. Even when I paid out for wire-care one year, Ma Bell did little to check my lines. The one tech who did come in informed me it was one of the phones. Well I've had many different phones over the years. After that one tech visit, they kept telling me everything was fine and there has to be "internal wiring" problem. Yeah, like I know that, and can you fix it, as I did pay for the service.

With VOIP, audio quality on my end has been suberb, except some callers did have some audio issues which I have related many times here.

Well this guy made three trips, the big one was 2 and half hours to rewire two jacks or $330 in cash. :huh:

Well need to call someone else, the first guy I called who offered $75 Flat, but wasn't readily available when I had no line.

This VOIP is costing me more, maybe not in the long run, but I'll be dead before.

Re: [Other] VOIP.MS

Wed, 08 Apr 2009 08:57:00 EDT

PX Eliezer : I'm puzzled. It sounds like your home had inherent phone wiring problems that would have been an issue with POTS or Voip.

And did you try to bargain and/or shop around for the rewiring services? In this economy $ 500 sounds like an awful lot.

Re: [Other] VOIP.MS

Wed, 08 Apr 2009 08:04:10 EDT

hockeynomad : Hmm, I did notice the Demarc box outside was open. Its door is not very secure and can open.

I did not issue a ticket with Bell. Have to check for any obvious loose wiring but my expertise ends there. I spent up to $500 paying a guy to reconfigure wiring to VOIP.

So much for my VOIP savings.

Re: [Other] VOIP.MS

Wed, 08 Apr 2009 00:38:53 EDT

Mango : Okay. How are the four dead jacks connected to the demarc? (Edit: I should have said "How SHOULD they be connected?") Most popular is the "home run" or "star" technique whereby a cable runs from each jack to the demarc.

Less popular for obvious reasons is the "daisy chain" technique whereby a cable runs from the demarc to the first jack, and then to the second jack, and then to the third jack...

I'd start by investigating the demarc. Any loose wires? How did the issue happen? Is the demarc outside, and is there any possibility the Telco came by and messed something up, in an effort to disconnect an old landline?

m.

Re: [Other] VOIP.MS

Tue, 07 Apr 2009 22:43:50 EDT

hockeynomad : Four of five jacks are dead except the one that is a demarc point. This one the ATA is connected to.

Re: [Other] VOIP.MS

Tue, 07 Apr 2009 20:48:51 EDT

Mango : I can't really figure out what the symptoms are from your post. Is it that all the jacks are dead, but you can plug a phone directly into the ATA and it will work?

Re: [Other] VOIP.MS

Tue, 07 Apr 2009 19:49:45 EDT

hockeynomad : Having interesting issue. Currently using phone answer machine for voicemail. Nothing going to this answer, apparently problem is that phone jack is dead.Well all jacks are dead lines except the DM point jack.

Damn I never knew VOIP would be a pain in the ass just to save a few bucks.

Anyway, nNow nothing is going to voip.ms message system. Live chat advised to change dial out setting of DID from 60s to 30s.

Just did that, will see what happens.

Any other ideas? Using Linksys 2102 ATA device.

Re: [Other] VOIP.MS

Sat, 04 Apr 2009 23:32:13 EDT

otty : Just noticed I had lost registration to ca2. Changed to us4 and it registered fine. My Nokia E51 doesn't have any connection logs so I can't tell when I lost registration. Anyone having problems? Anyone have a connection log to ca2?

EDIT: Nevermind. I think it was a settings error. I'm reconnected to ca2.

Re: [Other] VOIP.MS

Sat, 04 Apr 2009 19:55:52 EDT

wirelessbry : Hi Folks...

I thought you might be curious. I asked a few weeks ago about setting up VOIP.MS at home. I had a ton of problems initially. Someone on this blog suggested my router / ATA might be crap.

Anyhow, long story short, my Belkin Pre-N router was the issue. I put in place a standard WRT54G and it works beautifully. I did some digging and found out that Belkin are not always voip friendly.

Just wanted to thank all those that helped me out on the thread.

Re: [Other] VOIP.MS

Tue, 31 Mar 2009 17:47:25 EDT

WildChild : They were supposed to contact me today about the problem... I'm still waiting. This is not funny after two months...

Re: [Other] VOIP.MS

Mon, 30 Mar 2009 23:27:54 EDT

WildChild : My dial plan is good, the fast busy is on inbound calls, is intermittent and it only happens with calls from a particular provider.

Re: [Other] VOIP.MS

Mon, 30 Mar 2009 23:21:39 EDT

mazilo :

said by WildChild :

The fast busy tone problem I have since two month to contact my DID from Videotron is still not solved...

Fast busy tone is mostly caused by wrong dialplan strings, AFAIK.

Re: [Other] VOIP.MS

Mon, 30 Mar 2009 22:35:03 EDT

WildChild : The fast busy tone problem I have since two month to contact my DID from Videotron is still not solved... I asked to port my number from Videotron to voip.ms but with this problem, I'm considering cancelling this request because I don't want my friends/family, that are all with Videotron, not be able to contact me... This is very sad... This kind of situation (over two months to solve a problem) shouldn't happen. By the way I'm also not happy at how Videotron is dealing with the situation. I also contacted them two months ago and I'm also waiting after them...

Re: [Other] VOIP.MS

Sun, 29 Mar 2009 01:04:12 EDT

Mango : An email I received tonight:

Dear Customer,

We'll perform an emergency maintenance today at 1:00 AM EST on our database servers to improve performance en ensure continued service on the customer portal.

This will affect your access to the customer portal interface. Estimated time for completion is 2.5 hours. During this time you will not be able to access the customer portal.

VoIP servers are NOT affected by this maintenance, only web access to the customer portal.

Thank you for your understanding.

VoiP.ms Team

Re: [Other] VOIP.MS

Sat, 28 Mar 2009 18:47:16 EDT

Mango : Number porting is back online.

Re: [Other] VOIP.MS

Fri, 27 Mar 2009 08:19:24 EDT

hockeynomad : Just got a nifty tip from mlord of another voip.ms discussion in the the digital forum. Havn't tried it yet but I would like to block those annoying telemarketer calls.

"If you have a PAP2T for your VoIP connection, then you can program that device to deal with the annoying incoming call based upon its CID number.

This feature is under the User 1 and User 2 tabs, and allows setting automatic call-forward for specific incoming CID patterns. EDIT: just leave the forwarding number field blank, and the PAP2T will ignore/not answer the call. Or forward them back to themselves, so they don't tie up the line while being ignored.

Another nifty feature of the PAP2T, is the Caller ID Method, on the Regional settings page. If you're using DECT handsets, you can probably change this setting to ETSI FSK, which shows the calling ID information *with* the first ring, instead of just before the second ring."

Re: [Other] VOIP.MS

Thu, 26 Mar 2009 12:25:45 EDT

otty : I have actually had this happen from the beginning (2-3 months ago) but I never really thought of it as a big deal as it doesn't seem to affect the call at all. It's just very odd.

By the way, I assume you mean ringing tone not dial tone right?

Re: [Other] VOIP.MS

Thu, 26 Mar 2009 08:25:21 EDT

hockeynomad : Hey otty, I thought I was having a senior moment but I got the low tone european ringing tone as well.

Re: [Other] VOIP.MS

Tue, 24 Mar 2009 16:40:22 EDT

otty : I've had it happen now with 2 number in the 306 (Saskatoon) area code and I've had it in 250-837-**** interior BC (Revelstoke) area code.

Re: [Other] VOIP.MS

Tue, 24 Mar 2009 16:32:07 EDT

Mango :

said by otty :

I'm getting a strange issue using ca2 premium routing: when I call instead of hearing the normal ringing sound there is a low tone that pulsates like the ringing tone would but sounds more like it would in some European countries....

Not happening for me. Does this happen with any particular area code?

Opened call Friday that my new shiny DID doesn't pass echo test.

Perhaps I'm having a dense moment, but what does it mean when a DID doesn't pass an echo test? Do you mean there's lots of echo on incoming calls, with this DID only?

m.

Re: [Other] VOIP.MS

Tue, 24 Mar 2009 16:20:05 EDT

edmidor :

said by priller :

Doh! In voip.ms time, a few hours really means next week some time. ;)

Yep! Opened call Friday that my new shiny DID doesn't pass echo test. No reply... Tried to call support - their line was so noisy I couldn't hear the rep... I'm slowly getting concerned about all that...

Re: [Other] VOIP.MS

Tue, 24 Mar 2009 16:08:43 EDT

otty : I'm getting a strange issue using ca2 premium routing: when I call instead of hearing the normal ringing sound there is a low tone that pulsates like the ringing tone would but sounds more like it would in some European countries....more like a low beep. The call connects as normal but this is strange.

I've had it happen before just briefly after which it would ring like normal, but now at least at this one number it always happens. Anyone had this happen?

Re: [Other] VOIP.MS

Mon, 23 Mar 2009 13:58:46 EDT

priller :
Doh! In voip.ms time, a few hours really means next week some time. ;)

Re: [Other] VOIP.MS

Mon, 23 Mar 2009 13:54:38 EDT

Mango : When I hit 'Start Procedure' I get "We're finalizing the final details of the new porting procedure - You will be able to start new port procedures and consult details on ports in progress in a few hours - Thanks for your patience."

Re: [Other] VOIP.MS

Mon, 23 Mar 2009 12:15:06 EDT

priller :

said by stoked :

DID portability has been down for a week. :( Been trying to get my number ported should have done it sooner.

It's alive again.

Re: [Other] VOIP.MS

Sat, 21 Mar 2009 21:33:53 EDT

dpoliteski : I remember over a month ago someone in this forum from VOIP.ms said that there were new features (CID name database) and others to be rolled out shortly. This was also to have happened already but no news on the site. Any updates anyone else got?

Re: [Other] VOIP.MS

Fri, 20 Mar 2009 14:04:28 EDT

Mango : That's nothing. From their "Latest News" page:

"Regarding Interactive Voice Responses and Calling queues, we've been delayed but are working hard at it to bring these new features online this week, by Friday 27th." This has been there for weeks. They meant FEBRUARY 27th!

(The original post date is 2009-01-12 but it's been edited to say the above since then.)

m.

Re: [Other] VOIP.MS

Fri, 20 Mar 2009 13:48:31 EDT

priller : They should be very embarrassed that the message is still "The new porting system is being pushed online and is scheduled to be back online today, March 16th."

Today is March 20th!

Re: [Other] VOIP.MS

Fri, 20 Mar 2009 13:37:15 EDT

stoked : DID portability has been down for a week. :( Been trying to get my number ported should have done it sooner.

Re: [Other] VOIP.MS

Wed, 18 Mar 2009 22:18:24 EDT

Mango :

said by PX Eliezer :
Just curious, do you mean [e911 testing is] frowned upon in your city, the province, or all of Canada?

I only have information about my city; I'm not sure about the rest of Canada. I've got a colleague who refuses to dial 911 because of a rude response an operator gave her when she called about a bunch of guys shooting up in our back entrance - the op claimed it wasn't an emergency and refused to deal with it. I suspect they're trying to prevent the "My TV doesn't get channel 37!" calls although it's my opinion that making the general public afraid of calling 911 is a poor practice.

I have heard of people in the US calling and saying "This is a 911 test. Please read my address information to me so I can verify that it is correct." The operators apparently have been quite happy to do that. Dunno what the issue is...usually us Canadians are polite :P

m.

Re: [Other] VOIP.MS

Wed, 18 Mar 2009 21:27:09 EDT

priller :

said by Mango :

By the way, how long has the porting system been offline now? It's been scheduled to be back online "today" for a while.

I noticed it was unavailable late last week.

Re: [Other] VOIP.MS

Wed, 18 Mar 2009 21:20:32 EDT

PX Eliezer :

said by Mango :

It allowed me to set up e911 for Vancouver though I've never tested it as 911 testing is very frowned upon here. I'd be curious to hear from a Canadian who's used it.

My last provider advertised e911, and their website claimed my account was set up correctly, but when I had to use it, it was still only basic 911.

Just curious, do you mean frowned upon in your city, the province, or all of Canada?

Considering the problem last year with Comwave and the 911 crew in Ontario trying to answer a Voip 911 call in Alberta, it's surprising if they are so opposed to testing.

Re: [Other] VOIP.MS

Wed, 18 Mar 2009 21:13:47 EDT

Mango : By the way, how long has the porting system been offline now? It's been scheduled to be back online "today" for a while.

Re: [Other] VOIP.MS

Wed, 18 Mar 2009 21:11:29 EDT

Mango : It allowed me to set up e911 for Vancouver though I've never tested it as 911 testing is very frowned upon here. I'd be curious to hear from a Canadian who's used it.

My last provider advertised e911, and their website claimed my account was set up correctly, but when I had to use it, it was still only basic 911.

m.

Re: [Other] VOIP.MS

Tue, 17 Mar 2009 21:23:58 EDT

edmidor : Just tried to setup e911 on voip.ms, and it seems to offer only US option. voip.ms offers e911 in Canada, doesn't it?

Re: [Other] VOIP.MS

Sat, 14 Mar 2009 17:14:14 EDT

PX Eliezer : Most basically, does your phone have volume controls??

Otherwise, it may be an issue of these settings:

(FXS) Port Input Gain:
How loud you sound to others.

(FXS) Port Output Gain:
How loud others sound to you.

Should be in your manuals.

Re: [Other] VOIP.MS

Sat, 14 Mar 2009 17:02:13 EDT

hockeynomad : Changed some settings as per advice here on Linksys SPA 2102 and one byproduct since has been reduced volume on calls.

How to adjust?

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 23:15:40 EDT

otty : Yup, finally updated.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 23:08:21 EDT

WildChild : Same here, the CDR takes more than 1h to get updated, but it is.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 22:55:52 EDT

bthornhill : I noticed the CDR is taking far longer than the advertised one minute to update. All calls seems to appear, but upwards of an hour later.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 22:22:10 EDT

otty : I just made two long calls and they are not showing up on my call usage report (15 minutes later). Anyone else notice this? Looks like I got some free calls!

EDIT: Still going...4 calls now and still nothing showing up on my call detail records, nor is my balance going down. Time to call Antarctica? ;)

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 17:33:04 EDT

WildChild : The problem with the 20 secs delay before the telephone rings on the value route seems to be solved. I switched back to value route on sip.ca1.voip.ms and it works normally!

EDIT:
It seems that the DTMF problem affects more the value routing than the premium routing. The best combo right now for me seems to be Premium routing/INFO DTMF.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 17:30:16 EDT

WildChild :

said by hockeynomad :

interesting, last night an outgoing call had no rings, but the other side it did ring 4 X.

It happened to me a few times, don't know what can cause it. When it happens, I hears some white noise with the partern of the rinback tone. The telephone at the other end rings normally.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 17:23:42 EDT

WildChild :

said by priller :

Ah ... what port does your provider actually see you coming from after being NAT'ed. Both registered devices pictured above are sourcing from 5060, but after the router/firewall the port has now changed.

As I understand, the NAT router itself assign another public port to the device, even if the device's port in the local network is 5060, as the public port 5060 was probably used for another device, right?

said by priller :

Agree. And that's the reason for have a low enough registration interval .. to keep the connection in your router/firewall open.

Fair enough, if the router is rebooted and loses all his mappings, within 3 minutes they will be opened again. Also, if the registration expires quickly enough, the VoIP provider will route the calls to the voicemail faster when the device becomes missing. I like the idea.

said by priller :

Sure 1 or 2 devices can be managed ... but, I'm talking about scaling well beyond that. Think just a small SOHO with 10 IP Phones ... are you really going to try managing port forwarding for that and beyond?

As I said, port forwarding is only manageable with a small local network (read home network)

said by priller :

Over the years I've seen this many times. I'm not talking administrator issues. Something like a VPN client provided to an employee by their corporate IT comes home and tries to connect to the office. It never negotiates because port forwarding has been set up and the appropriate packets get "hijacked". If the unnecessary port forward had not been done, there never would have been an issue.

How would UPnP work with VoIP? Would it be a good solution? A device registers to the router and ask it to automatically forward the necessary ports. KeepAlive packets can be sent less often. Many software/devices work like this to open ports for incoming connections (XBOX, BitTorrent, MSN Messenger, ...). They prepare the router to receive incoming packets without having to keep a connection opened by sending outgoing packets from time to time. Maybe UPnP isn't well suited for enterprises?

said by priller :

No. This is unrelated to device provisioning, this is part of the registration process and the negotiated parameters.

Thank you for the information, I wasn't aware of it.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 16:28:06 EDT

hockeynomad : interesting, last night an outgoing call had no rings, but the other side it did ring 4 X.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 15:47:21 EDT

Mango : Packet loss issue to CA2 appears to have been solved.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 15:37:05 EDT

priller :

said by WildChild :

Anyway, isn't the 5060 port assigned to SIP?

Ah ... what port does your provider actually see you coming from after being NAT'ed. Both registered devices pictured above are sourcing from 5060, but after the router/firewall the port has now changed.

said by WildChild :

About b), an incoming call could be seen line an unsolicited packet if it comes after a lot of time, port forwarding in this case can be OK.

Agree. And that's the reason for have a low enough registration interval .. to keep the connection in your router/firewall open.

said by WildChild :

I don't consider forwarding this particular port to my ATA a problem in this case. No problem either for 5061 assigned to my 2nd line on my ATA (default port for the 2nd line in Linksys SPA2102).

Sure 1 or 2 devices can be managed ... but, I'm talking about scaling well beyond that. Think just a small SOHO with 10 IP Phones ... are you really going to try managing port forwarding for that and beyond?

said by WildChild :

About the VPN, using this kind of services on port assigned to other software can always cause problems. It's up to the administrator to correctly configure/install his software/network.

Over the years I've seen this many times. I'm not talking administrator issues. Something like a VPN client provided to an employee by their corporate IT comes home and tries to connect to the office. It never negotiates because port forwarding has been set up and the appropriate packets get "hijacked". If the unnecessary port forward had not been done, there never would have been an issue.

said by WildChild :

Just a question, are you sure the VoIP provider can change the registration delay? If I set it to 15 secs, with provisioning disabled on my device (BYOD), the provider has no control on when the device will register, right?

No. This is unrelated to device provisioning, this is part of the registration process and the negotiated parameters.

Even on my SPA9000 IP PBX, I can set "Reg Min Expires" and "Reg Max Expires" times that will override what the registering device has in it's configuration, during the registration process.

To give an example of the opposite ... My line to Vitelity is set to 180 seconds. However, when I register, they force that down to 60 seconds. I cannot change that.

--

Good discussion.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 14:11:46 EDT

WildChild :

said by priller :

Fair enough.
b) Port Forwarding Breaks Things - When people forward a broad range of UDP ports to their ATA / IP Phone, you run a high risk of breaking other things. The most common is breaking VPN. Somebody tries to establish a VPN to their office and it doesn't work. Why? Because the VPN is trying to use the same UDP ports that you are forwarding to the ATA and they never make to where they are supposed to go. The intent of Port Forwading is to allow unsolicited packets to come into your network (e.g. a web server). If you are using a VoIP provider you register to, all traffic is originally initiated by the ATA / IP Phone. Even when you have an incoming call, the INVITE from the provider is just return traffic to an already established connection. You just have to make sure your timers keep that open.

c) Port Forwarding Does Not Scale - If you are forwarding all this traffic to one device, what are you going to do if you add additional ATA / IP Phones???? Hummmm

About b), an incoming call could be seen line an unsolicited packet if it comes after a lot of time, port forwarding in this case can be OK. Anyway, isn't the 5060 port assigned to SIP?

»www.iana.org/assignments/port-numbers

I don't consider forwarding this particular port to my ATA a problem in this case. No problem either for 5061 assigned to my 2nd line on my ATA (default port for the 2nd line in Linksys SPA2102).

About the VPN, using this kind of services on port assigned to other software can always cause problems. It's up to the administrator to correctly configure/install his software/network.

About c), you are totally right. But in my case, it's a small home network with a few devices on the router. I can easily track them all and I have no problem to remember what port is forwarded to what device/computer. If I have to add another VoIP ATA. I can just configure the ATA to use another incoming port and I'll forward it. I understand this is not well suited with a NAT in a big corporation with hundreds of VoIP devices to manage.

Just a question, are you sure the VoIP provider can change the registration delay? If I set it to 15 secs, with provisioning disabled on my device (BYOD), the provider has no control on when the device will register, right?

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 13:59:31 EDT

Mango : I'm seeing 7% packet loss to the CA2 server today. I also see 7% packet loss to my last hop before CA2, cn45-core.mountaincable.net [24.102.5.18], but not the second-to-last. »just-ping.com/ confirms this.

I have reported it to support and switched to CA1 in the meantime.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 10:29:31 EDT

priller : Fair enough.

There shouldn't be a need to forward ports for the RTP. When a call is being setup, the server and ports to use for the RTP (voice) packets is contained in the SIP/SDP message. The ATA / IP Phone will initiate the RTP stream to this server. Therefore, the router/firewall will allow the incoming RTP , because it is just return traffic.

Just some other items for all to consider:

a) The provider can set the minimum registration interval. For example, if you set your registration interval to 60 seconds and the provider only wants people re-registering at 180 seconds, your 60 seconds will be overridden. You will only register at 180 seconds regardless of the fact that you have set it to 60.

b) Port Forwarding Breaks Things - When people forward a broad range of UDP ports to their ATA / IP Phone, you run a high risk of breaking other things. The most common is breaking VPN. Somebody tries to establish a VPN to their office and it doesn't work. Why? Because the VPN is trying to use the same UDP ports that you are forwarding to the ATA and they never make to where they are supposed to go. The intent of Port Forwading is to allow unsolicited packets to come into your network (e.g. a web server). If you are using a VoIP provider you register to, all traffic is originally initiated by the ATA / IP Phone. Even when you have an incoming call, the INVITE from the provider is just return traffic to an already established connection. You just have to make sure your timers keep that open.

c) Port Forwarding Does Not Scale - If you are forwarding all this traffic to one device, what are you going to do if you add additional ATA / IP Phones???? Hummmm

d) STUN is nearly obsolete - That was the original NAT solution. These days, either your router should have a SIP ALG that will rewrite the packets accordingly or the provider does a fixup on their SBC. If they aren't doing that, shame on them.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 09:38:26 EDT

WildChild : Oh, true, my registration is set to 3600 secs. If I remember, I set the timeout on my router to 5 minutes. I don't want to overload the server with too many registrations and I think it's safer to have the port forwarding enabled at least for the SIP ports (not sure if it's necessary for the RTP ports). This way, if the router decides to cut the pipe early for any reason, the port forwarding will kick take care of routing the packets correctly.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 09:01:36 EDT

priller :

said by WildChild :

If I define no port forwards, after 5 minutes, when my router drop the UDP IP/port association, my ATA won't receive any call.

What's your registration interval? Should be no higher than 180 seconds.

That's a classic symptom of your router/firewall timing out the connection. You just have to have the registration interval set low enough to keep it open. Then you should be able to get rid of the port forwarding and most likely STUN.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 08:59:21 EDT

WildChild : Oh, port forwarding can be necessary... In my case it was and I now have a stable connection with the server without using a STUN server. If I define no port forwards, after 5 minutes, when my router drop the UDP IP/port association, my ATA won't receive any call. STUN helped to keep this IP/port association cached in the router by sending a KeepAlive before the router dropped it but I don't like to rely on a 3th party server.

About my DTMF problem, it happens on the premium and on the value route.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 08:55:48 EDT

WildChild : All my settings are already like this, latest firmware. The ATA recognize correctly the digits but they are not all received at the other end unless I press them long enough and I give a small delay between every digits.

With InBand, no problem with my work's voicemail system but the 4747 DTMF test doesn't work.

It seems only the INFO mode gives "reliable" results every times.

said by hok :

In your ATA, looking for following settings and try them please:
FXS Port Input Gain: -3
FXS Port Output Gain: -3
DTMF Playback Level: -16
DTMF Playback Length: .1
Detect ABCD: yes
Playback ABCD: yes

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 08:53:11 EDT

priller :

said by wirelessbry :

If that doesn't work, I think I'll go and buy a new linksys 3102. It's funny you mention the ATA. When I first tried it, I couldn't get Port 1 to work and then tried Port 2 and it did work. So, maybe the SPA 2000 is a little flaky after these years given its age.

Just as a point of reference ... I currently use or have used the following on voip.ms. I use DTMF for conference calls on a daily basis and not once have I ever experienced a DTFM problem (premium routes).

SPA9000
PAP2T
SPA942
SPA1001
Polycom IP 550

So, maybe that SPA2000 is a bit "off".

Also, please, no port forwarding ... totally unnecessary.

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 08:32:04 EDT

hockeynomad : Maybe time for Callcentric................

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 06:45:05 EDT

wirelessbry : I'm going to try solving the problem in two ways.

1) Remove the router completely to see if its the problem in general

2) Swapping the router with a Linksys. I've been trying to get port forwarding on the 5060 and the RTP ~13600 range to no avail.

If that doesn't work, I think I'll go and buy a new linksys 3102. It's funny you mention the ATA. When I first tried it, I couldn't get Port 1 to work and then tried Port 2 and it did work. So, maybe the SPA 2000 is a little flaky after these years given its age.

Thanks for the ideas. I'll let you if they work.

bryan

Re: [Other] VOIP.MS

Thu, 12 Mar 2009 06:02:59 EDT

hok : In your ATA, looking for following settings and try them please:
FXS Port Input Gain: -3
FXS Port Output Gain: -3
DTMF Playback Level: -16
DTMF Playback Length: .1
Detect ABCD: yes
Playback ABCD: yes

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 22:10:31 EDT

WildChild : Hum, it seems that DTMF problems are common with VoIP! I have to push on the keys very slowly and leave some time between key press so my work's voicemail system recognize what was entered. What are your suggestions about DTMF problems?

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 19:53:40 EDT

bthornhill :

said by WildChild :

Currently on chat with voip.ms. They were able to reproduce the delay issue. Only present on value route. A switch to the premium route can be done as a temporary measure to solve the problem.

Thank-you! That's great news. I just tested it, and the family reported that the delay went from 30s to about 10s. The 10s might well be because they were calling me on mobile and I'm roaming in the USA currently.

I was planning to switch to premium routing anyways to try and mitigate echos when I call western Canada.

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 16:59:39 EDT

WildChild : Currently on chat with voip.ms. They were able to reproduce the delay issue. Only present on value route. A switch to the premium route can be done as a temporary measure to solve the problem.

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 16:24:44 EDT

bthornhill :

said by wirelessbry :

Question: Anyone in Toronto getting materially faster ping times? Who is your ISP. (And... if you perform a traceroute... are the hops staying in Canada??) I'm just trying to determine if the ping time is a function of carrier routing.

I get about 35ms to the Montreal server from the GTA. On Bell DSL.

The Toronto server gets about 50ms pings and routes me through the US with an additional hop versus Montreal.

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 16:23:29 EDT

bthornhill : Insofar as I understand, if you press the terminator (# in my case) after dialing, then the dial plan itself is unrelated to the problem. I get the 15sec delay regardless of whether or not I press the terminator or let the dial plan do it for me.

For the record, I am on the Montreal server, using the value routing.

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 16:10:34 EDT

WildChild : Do you use premium or value route?

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 16:09:58 EDT

WildChild : Hum, I still have problems with AVT... Only InBand seems to work with work's voicemail system.

said by WildChild :

I'm trying to access my work's voicemail system but it refuses me the access when I use my voip.ms line. Can this be related to the codec used (remote system too sensitive to subtle tone changes)? Has the DTMF setting, found on voip.ms control panel, something to do about this?

EDIT:
Changed the DTMF TX Method to AVT as suggested by voip.ms tech support. I tried InBand and it seems to work but I've read that it will only work with the G711 codec that doesn't compress the voice.

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 16:05:19 EDT

sream : I've been using pop us4 heavily the past 5 days without a hitch. You're having an issue with calls connecting if I'm reading correctly.

Re: [Other] VOIP.MS

Wed, 11 Mar 2009 15:56:13 EDT

WildChild : Hum, I now have the issue with sip.us4.voip.ms...

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 21:42:09 EDT

WildChild : No problem with sip.us4.voip.ms. Tried it while chatting with the tech support. They know it's a server problem.

said by WildChild :

Same behaviour here on sip.ca1.voip.ms or sip.ca2.voip.ms.

said by hockeynomad :

OK, now dialing out is taking over 30 seconds on ALL calls to engage, only a bit faster should I use (#).

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 21:08:49 EDT

priller :

said by wirelessbry :

I'm stumbling because I'm thinking that latency is my problem.

I simply can't get a clean outbound call without dropout. It's weird, when I make a test call, it's almost as if the call drops (no sound) every second or steamboat.

Is it maybe an ATA issue or router issue?

With your 50ms latency, are you still getting "good calls"?

bryan

There is absolutely nothing wrong with 50ms, or 100ms. 50ms is not even close to being problematic. Remember ... VoIP, specifically the RTP packets that carries the voice, is a streaming media. One packet is always following another. You don't send one packet and have to wait for a reply to send another. If you are experiencing audio drop outs, it is either due to 1) packets are being dropped in transit or 2) the depth of the jitter buffer needs to be increased.

And there is a possibility that the ATA itself has gone bad.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 20:24:01 EDT

WildChild : I'm trying to access my work's voicemail system but it refuses me the access when I use my voip.ms line. Can this be related to the codec used (remote system too sensitive to subtle tone changes)? Has the DTMF setting, found on voip.ms control panel, something to do about this?

EDIT:
Changed the DTMF TX Method to AVT as suggested by voip.ms tech support. I tried InBand and it seems to work but I've read that it will only work with the G711 codec that doesn't compress the voice.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 20:13:31 EDT

WildChild : Can you take this test:

»/linequality

??

Maybe your connection is losing packets. Not good for VoIP...

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 19:38:56 EDT

otty : I'm out in Saskatchewan so there is no server nearby. My pings to ca2 are 70-200ms. Quality is still fine. There is a bit of lag noticeable in an echo test, but nothing serious. I imagine your problem is not latency. Probably jitter or settings. Try another server.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 19:20:47 EDT

wirelessbry : I'm stumbling because I'm thinking that latency is my problem.

I simply can't get a clean outbound call without dropout. It's weird, when I make a test call, it's almost as if the call drops (no sound) every second or steamboat.

Is it maybe an ATA issue or router issue?

With your 50ms latency, are you still getting "good calls"?

bryan

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 19:16:18 EDT

garys_2k : I see 52 ms to sip.ca1.voip.ms from here in SE MI, too. My ISP hops me through Chicago to get there.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 19:09:14 EDT

wirelessbry : I wish I was getting 8ms as well.

So, to be honest, I'm not sure how to interpret these results. Request timed out doesn't seem too good!

The idea about removing all other network connections is a good idea... doing it now.

Bryan

C:\Users\Burt>tracert sip.ca1.voip.ms
Tracing route to sip.ca1.voip.ms [67.205.74.164]
over a maximum of 30 hops:

1 1 ms 1 ms 1 ms 192.168.0.1
2 3 ms 2 ms 2 ms 192.168.2.1
3 * * * Request timed out.
4 * * * Request timed out.
5 * * * Request timed out.
6 * * * Request timed out.
7 * * * Request timed out.
8 * * * Request timed out.
9 * * * Request timed out.
10 * * * Request timed out.
11 52 ms 52 ms 51 ms ip-67-205-74-164.static.privatedns.com [6
4.164]

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 18:09:12 EDT

WildChild : Here's the log:

Mar 10 18:07:27 Linksys [0]Off Hook
Mar 10 18:07:29 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:29 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:29 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:29 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:30 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:30 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:30 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:30 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:30 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:31 Linksys 2. Report digit x (1)(40 ms)
Mar 10 18:07:31 Linksys Calling:xxxxxxxxxx@sip.ca1.voip.ms:0
Mar 10 18:07:31 Linksys [0:0]AUD ALLOC CALL (port=16394)
Mar 10 18:07:31 Linksys [0:0]RTP Rx Up
Mar 10 18:07:52 Linksys [0:0]ENC INIT 0
Mar 10 18:07:52 Linksys [0:0]RTP Tx Up (pt=0->43cd4aa4:18246)
Mar 10 18:07:52 Linksys [0:0]RTCP Tx Up
Mar 10 18:07:52 Linksys CC:CallProgress
Mar 10 18:07:53 Linksys [0:0]RTP Rx 1st PKT @16394(2)
Mar 10 18:07:53 Linksys [0:0]DEC INIT 0
Mar 10 18:08:02 Linksys [0]On Hook
Mar 10 18:08:02 Linksys [0]FM Alert Stop RxTx (c=00241694;a=0)
Mar 10 18:08:02 Linksys [0:0]AUD Rel Call
Mar 10 18:08:02 Linksys DLG Terminated 29420c

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 18:05:39 EDT

WildChild : I have a syslog server already available and configured (I have a FreeBSD server always running). Did not thought about checking the logs. I'll test it right now.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:54:29 EDT

Mango : If any of you having the dial delay are able to set up a Syslog server I would be interested to see what your log says. I've used Kiwi Syslog Server which is free and easy to install. After installing it, I set the Debug Server on my ATA to the IP of my PC. If you have a firewall on your PC you may need to add a rule to allow Kiwi to act as a server.

After making a test call, it should spit out lots of data. Edit out anything confidential (personal phone numbers, username, etc) before posting.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:38:08 EDT

WildChild : I tried resetting my ATA to factory default with no success. This delay isn't caused by a setting I changed!

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:37:22 EDT

WildChild : I also get 10mbits up and down on Internet. Currently I'm a consultant in a minister of Government of Québec. My batch of consultants is on a network trunk with no outgoing port and access restriction. The internal employees are all behind a proxy and are jealous of us because they don't have speed, access to anything and their network trunk crashes when they download a file over 10 MB in size.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:36:39 EDT

priller :

No problem there.

Also, don't over analyze ping times. The differences you are looking at aren't going to matter in any way.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:22:28 EDT

Mango :

said by WildChild :
I get 8ms ping from my work

*drool!*

50ms ping is not great but not too bad. I see 70-90ms from Vancouver to the Toronto or Montreal server and 40ms to the Los Angeles server. I get crystal clear audio with both. True, 50ms is a bit long for Toronto to Toronto, especially since I can get from here to LA faster, but as far as I am aware there's not much you can do about ISP routing, other than moving in with WildChild.

Here's one way you can test to see if QoS would help you: unplug everything from your router except your VoIP device. Make a few internal test calls (try 4443, 811, or *98) and see if you notice the problem. In this case, your VoIP device is the only thing using the connection, so if you still have quality issues, they are not to do with bandwidth sharing and QoS would likely not solve the problem. If on the other hand everything works perfectly, then some other device is using too much bandwidth for your VoIP call and a WRT54GL with Tomato should improve things significantly.

Long ping times/hops shouldn't cause choppy audio unless there is lots of jitter (wide variation in ping times). If you start pinging the server and see something like 40ms, 50ms, 200ms, 40ms, 300ms, then you have a jitter problem and you should adjust the jitter buffer on your device appropriately. (And, call your internet provider to see if they'll fix it.)

One other contributor to choppy voice is an incorrectly set RTP packet size. The default for Cisco devices is 0.03; try 0.02 instead.

Edited to add a very curious traceroute:
Tracing route to sip.ca1.voip.ms [67.205.74.164] over a maximum of 30 hops: 1 2 ms 1 ms 1 ms 192.168.1.1 2 * * * Request timed out. 3 7 ms 8 ms 7 ms rd1bb-ge3-0-0-1.vc.shawcable.net [64.59.158.178] 4 23 ms 10 ms 8 ms rc2bb-tge0-15-0-0.vc.shawcable.net [66.163.69.137] 5 24 ms 22 ms 21 ms rc2ar-pos11-0-0.ed.shawcable.net [66.163.77.221] 6 26 ms 24 ms 23 ms rc1we-tge0-8-4-0.ed.shawcable.net [66.163.70.37] 7 41 ms 41 ms 39 ms rc2ch-pos14-0-0.il.shawcable.net [66.163.76.62] 8 72 ms 64 ms 71 ms rc1sh-pos15-0.mt.shawcable.net [66.163.76.221] 9 74 ms 67 ms 65 ms rc2sh-ge6-0-0.mt.shawcable.net [66.163.66.82] 10 65 ms 65 ms 65 ms ia-fnrt-bb02-ge1-1-1040.vtl.net [207.96.146.25] 11 72 ms 69 ms 72 ms ia-cnnu-bb04-pos11-0.vtl.net [216.113.122.25] 12 72 ms 72 ms 78 ms 216.113.123.18 13 86 ms 94 ms 75 ms te6-1.cl-core04.vtl.mtl.iweb.com [207.253.238.114] 14 74 ms 74 ms 71 ms ip-67-205-74-164.static.privatedns.com [67.205.74.164] Trace complete.

"il"??! Is Shaw seriously routing me from Edmonton, to Chicago, and back to Manitoba? I pinged the il router from a server in Chicago and it is only 10ms away. Please tell me I'm reading this wrong because that would be just silly.

m.

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:08:20 EDT

WildChild : Start, Execute, "cmd", "tracert sip.ca1.voip.ms". Paste here what it gives. I have 18ms ping to sip.ca1.voip.ms. My traceroute is the following:

Détermination de l'itinéraire vers sip.ca1.voip.ms [67.205.74.164]
avec un maximum de 30 sauts :

1 1 ms 1 ms 1 ms wasteland.XXXX.net [192.168.69.69]
2 * * * Délai d'attente de la demande dépassé.
3 10 ms 10 ms 9 ms 24.200.227.157
4 10 ms 9 ms 9 ms 24.200.250.41
5 11 ms 9 ms 9 ms 216.113.123.65
6 11 ms 9 ms 10 ms 216.113.123.10
7 17 ms 17 ms 17 ms te6-1.cl-core05.vtl.mtl.iweb.com [216.113.13.10]

8 17 ms 17 ms 17 ms ip-67-205-74-164.static.privatedns.com [67.205.7
4.164]

Itinéraire déterminé.

(Sorry for the french)

I get 8ms ping from my work. Fiber from Bell in Québec City

If your traceroute goes through NY, maybe it would be better for you to use sip.us4.voip.ms?

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:01:15 EDT

wirelessbry : Hello,

If anyone has any thoughts, I have two topics to ask...

1) Ping time. There has been a few comments on ping times. I am one of those users that has a ~50ms ping time to the MTL server. I'm amazed at the very short ping times that other users are achieving. When I perform a traceroute from my Bell DSL line, it's clear that the routing is going through the US first before getting back to the server in MTL.

Question: Anyone in Toronto getting materially faster ping times? Who is your ISP. (And... if you perform a traceroute... are the hops staying in Canada??) I'm just trying to determine if the ping time is a function of carrier routing.

2) QOS on the router. When I make calls now, they are somewhat choppy. I suspect that's a function of the longer hopping and longer ping times. (I'm currently on the premium setting.) My router doesn't have QOS.

Question: Do you need QOS (e.g., Tomato firmware on a Linksys WRT54) to optimize the voice packets for best results?

Anyhow, I'm currently using an older Sipura 200 hooked up to a Belkin Pre-N router.

Any thoughts, ideas or suggestions greatly appreciated.

Bryan

Re: [Other] VOIP.MS

Tue, 10 Mar 2009 17:00:51 EDT

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 22:55:54 EDT

Mango :

said by priller :
I would hope the support delays of January are behind us.

I was hoping for that too. I'm still waiting on a response from my email from March 2. I resent this on March 6. It's frustrating that the support is so hit-and-miss.

said by WildChild :
I don't want them to have trouble contacting me, or do I? :)

I wish some of my family used Videotron :D

m.

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 20:31:55 EDT

WildChild : Contacted voip.ms about the 30 secs delay. :)
On my side, I still have problems contacting my DID from Videotron's numbers... It has been 3 weeks since the problem has been reported. I hope it will be solved soon. I initiated a number porting process today to bring my home phone number from Videotron to voip.ms but all my family uses Videotron. I don't want them to have trouble contacting me, or do I? :)

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 18:54:44 EDT

priller :
I would hope the support delays of January are behind us. Two or three weeks ago I contacted support via email and received a response within an hour. I also have used chat and got an immediate informed response.

If you're having a problem like post-dial delay, nobody here can fix that for ya, so what do you have to loose?

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 16:45:21 EDT

hockeynomad : Have you contacted support?
============================================================

Ha. I did that once long ago and still waiting for response.

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 16:40:12 EDT

Mango : No problems here on ca1, ca2 or us3.

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 16:15:46 EDT

otty : No problems here on ca2.

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 16:05:29 EDT

priller :
sip.us2.voip.ms has been fine all day.

Problem isolated to "ca"? Have you contacted support?

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 15:57:43 EDT

WildChild : Same behaviour here on sip.ca1.voip.ms or sip.ca2.voip.ms.

said by hockeynomad :

OK, now dialing out is taking over 30 seconds on ALL calls to engage, only a bit faster should I use (#).

Re: [Other] VOIP.MS

Mon, 09 Mar 2009 11:04:23 EDT

hockeynomad : OK, now dialing out is taking over 30 seconds on ALL calls to engage, only a bit faster should I use (#).

CA1 and CA2 servers resolve to same IP address

Sat, 07 Mar 2009 14:44:15 EDT

bthornhill : So I was testing pings between CA1 and CA2.sip.voip.ms and I see they are resolving to the same IP address! What's with that?

Since there's no centralized email, wouldn't that mean that people being directed to the other server would have no messages?

EDIT: Pls disregard. Apparently ca1.sip.voip.ms is not the same as sip.ca1.voip.,s

Re: [Other] VOIP.MS

Sat, 07 Mar 2009 13:51:29 EDT

Mango : Fortunately, the port is already complete. I just wanted to find out the cause of the downtime, in case it was something that was my fault, so that I could avoid it for next time. I'm just being a Curious George as usual :)

m.

Re: [Other] VOIP.MS

Sat, 07 Mar 2009 13:34:56 EDT

hockeynomad : Mango, I know you are really anxious to get your port. I see you in two threads.

I would advise you message or PM Marin personally to expedite your port. That's what I did. Check page 2 and 3 of this thread.

It should get your DID that much faster.

Re: [Other] VOIP.MS

Sat, 07 Mar 2009 11:25:06 EDT

Mango : Martin,

Do you know what went wrong with my number port? FOC was February 24, but nothing happened until March 2 when I got an email from my office saying my phone number was not in service. I reported it to support who said they would open a ticket with the carrier, and then the port went through about six hours after I noticed the issue. (9 weeks 2 days.)

I replied to the ticket twice and asked, on March 2 and March 6, but I have not received a response to either of those.

m.

Re: [Other] VOIP.MS

Fri, 06 Mar 2009 09:59:57 EDT

sream : Ok, well the faq isn't visible when you're logged in. Which is kind of silly.

So far on call quality I give it 5/5. Did a bit of an extreme test.

Out on VoIP.ms --> ipkall did ---> callcentric (my catchall) ---> my pbx call attendant which forwards out ---> VoIP.ms ---> cell phone. Voice quality was very good. Of course there was a nice delay with all those hops.

Re: [Other] VOIP.MS

Thu, 05 Mar 2009 17:45:52 EDT

Mango : What sort of documentation, sream? If you have a question, perhaps we could help? :)

Re: [Other] VOIP.MS

Thu, 05 Mar 2009 06:46:47 EDT

sream : Well I signed up. Getting some test did's and I'm going to start routing outbound to voip.ms. Voice quality is what I'm used to which is good. You guys should really really get some documentation up though.

Re: [Other] VOIP.MS

Wed, 04 Mar 2009 18:29:06 EDT

PX Eliezer : This thread hopefully can help:

»Re: sip broker pstn number to voip.ms sub-account number. how?

I do wish that Voip.MS supported SipBroker. Currently there is no outbound support at all. And as you saw, inbound support has issues.

A provider that has very good peering with SipBroker (and peering in general) is CallCentric.

Re: [Other] VOIP.MS

Wed, 04 Mar 2009 16:50:52 EDT

AndrewZ :

said by gsar :

The voip.ms sipbroker peering code (*8570) also now seems to work.

Doesn't work for me, with either main account or subaccount.

Voip.ms support replied "We have no support for a code such as *8570 right now."

Any comments/suggestions?

Re: [Other] VOIP.MS

Wed, 04 Mar 2009 11:32:48 EDT

hockeynomad : Live chat this morning with Stephen:

Stephen Gutierrez: Im sorry sir, CallerID Filter is not available right now, we're planning to release it soon though.

Hockeynomad: When is soon? ETA we were told was mid-Feb?

Stephen Gutierrez: yes sir, but, we had some delays on the release of the PBX features, and that's why this feature had been delayed too, we need to make sure that all this feature work 100% before we release it, right now our tech are in testing phase, we hope that on the end of this week it's done

Re: [Other] VOIP.MS

Wed, 04 Mar 2009 08:56:59 EDT

priller : Posted on 2/6:

said by MartinM :

I will provide some updates regarding Incoming features for VoIP.ms, as requested by many customers.

Calling Queues, IVR (Digital Receptionist) and Recordings are currently in testing phase. This mean it's ready for a launch in the next few days. (Eta 1 week at most for launch)

A big change regarding CNAM Queries: We're reworking the system to cache your queries on a 24 hours basis. That mean that if "John Smith" Calls you 5 times in the day, you will be charged for only 1 Query. (Eta 14 days)

CNAM Directory / Speed DIal: You will be able to create entries with specific number. When you will receive an incoming call, if the CallerID Number matches one of the entries you have programmed, no Querie will be made to the CNAM/LIDB repository. It will instead display the name you have programmed for that entry. Eta: 15-25 Days

CallerID Blocking: You will be able to block Anonymous Calls and Specific CallerID Numbers (Eta 1-2 weeks)

Features codes: Most of the features that are accessible via the web interface, will have feature codes added to them. This will allow you to enable/disable features by dialing specific codes without havingto log in to the interface. I do not have a list of the specific features that will be added via this, but most standard features will be added (Last Caller, block last caller, Activate forwarding etc). I will provide the list when we are done with the planning. (Eta: 4-6 weeks)

Inbound DID Fail over according to condition: At the moment, fail over, if enabled, is activated on the following conditions: Busy or Unreachable. We'll add custom fail over conditions soon, that will let you program different routes according to the 3 following conditions: No Answer, Line Busy, Device or PBX Unreachable

I would like to thank everyone for their suggestions. Please keep them coming, we appreciate every suggestions and try to integrate the majority of them.

Martin .... It's been almost 4 weeks since you provided this list and none have appeared. Any updates for us?? Thanks!

Re: [Other] VOIP.MS

Tue, 03 Mar 2009 11:23:40 EDT

Mango : Wasn't me :D I like Caller ID too much. Oooo, and seven servers...I could have a different server for every day of the week if I wanted!!

What? Well, I just figured, my wife has different shoes for every day of the week, can't the equivalent for me be to have a server for every day of the week? Ok, I'll stop talking now.

m.

Re: [Other] VOIP.MS

Tue, 03 Mar 2009 08:06:32 EDT

hockeynomad : Mango, I thought you were going with Callcentric?

I'm sticking with voip.ms. Despite some issues, can't beat voice quality and price.

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 20:44:00 EDT

Mango : I'm back up and running. Looks like my number port has now gone through.

m.

Saskatchewan Voipers left out in the cold.

Mon, 02 Mar 2009 20:32:33 EDT

PX Eliezer : News story---2/27/2009

REGINA -- SaskTel’s WebCall customers only have one day left before their Internet-based telephone service is permanently disconnected.

WebCall, a voice-over-Internet protocol (VoIP) service by Navigata, a member of the SaskTel group of companies, will be terminated on Sunday after providing almost five years of service.

While WebCall service was available in B.C., Alberta and Saskatchewan, Michelle Englot, spokeswoman for SaskTel, said there were only about 130 WebCall customers in Saskatchewan, and approximately 270 outside of the province.

“It had limited market success and had been operating at a significant loss as a result,” she said.

WebCall customers were informed of the planned termination in January, and were told to find an alternative solution prior to the March 1 deadline to avoid a disruption in service.

For in-province customers wishing to keep or transfer their existing VoIP phone number, options on WebCall’s website (www.webcall.ca) varied depending on whether customers lived in WebCall’s three exchanges — Regina, Saskatoon, or Prince Albert.

Customers within the three exchange areas who wish to keep their number have the option of transferring their service to SaskTel local access, mobility service, or to a competitor. However, they must live and stay within the exchanges.

Those outside the exchange areas can also choose to transfer the service to SaskTel local access, but cannot keep their current phone number.

If people want to receive VoIP service from a competitor, Englot said SaskTel will honour any porting request that comes to the company, as is required by the CRTC.

“We’re more than willing to work with any service provider who wants to port numbers,” she said.

While it sounds easy enough to switch to a competitor, it is up to the alternate company to request porting WebCall numbers. Englot said that because VoIP companies are often resellers, unlike local exchange carriers, they aren’t required to port numbers.

“It’s up to (customers) to evaluate the competitors that are out there,” said Englot. “We don’t know who accepts ported numbers or not.”

Andrew Day, senior vice-president of residential services at Primus, said it would be impossible to schedule the porting of a number this close to the March 1 deadline. However, Day had a suggestion to help WebCall customers keep their phone numbers.

“If SaskTel would hold the number for a WebCall customer … then give us that number, we could set up a customer as a new customer,” he said. “So it’s a very manual way for the customer to keep their existing number.”

However, Day said this could only work with the company’s three exchanges within the province’s 306 area code — numbers with the prefixes 523, 564, and 566.

Englot said she didn’t know which prefixes WebCall customers used, but said if a company was interested in doing this, it should contact SaskTel’s carrier services group.

SaskTel’s WebCall customers only have one day left before their Internet-based telephone service is permanently disconnected.

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 19:54:58 EDT

nitzan : More likely cause - not enough competition and not enough pro-consumer regulation.

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 19:48:48 EDT

PX Eliezer : Considering that you are going on more than 9 weeks for a port, and are still waiting.....

Quite seriously, the least they can do is waive the fee for the port. And maybe some additional account credits too. :(

------------------------------------

Why do Canadian port requests take so long? Is it because in the cold winter, the ports in Canada are frozen?

(Admittedly bad joke, folks, sorry.) :p

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 18:55:09 EDT

PX Eliezer : That's certainly troubling.

See also:
»What is a FOC when porting a number?

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 18:39:15 EDT

Mango : I got an email from a colleague at 1PM pacific saying that my DID was not working. Indeed, when I tried it, I got the "We're sorry; this number is not in service."

The FOC for this DID to port in to VoIP.ms was February 24, but it hasn't ported in yet. (I thought the FOC date was the date at which point the DID would be usable on VoIP.ms? Maybe I'm wrong?) I used Live Chat and they say they will open a ticket with their carrier.

At 9 weeks 2 days.

m.

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 14:22:51 EDT

priller :

said by wuliu7 :

I could not find from the website. How much does it cost to port a number? I am in US.

USA: $25 One-Time fee, Usually complete in 2-4 weeks.
Canada: $25 One-Time Fee, Usually complete in 3-8 week.
Toll-Free: $25 One-Time fee. USA: 2-4 Weeks, Canada: 3-8 Weeks

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 14:16:09 EDT

wuliu7 : I could not find from the website. How much does it cost to port a number? I am in US.

Thanks.

Re: [Other] VOIP.MS

Mon, 02 Mar 2009 12:03:00 EDT

bthornhill :

said by bthornhill :

said by MartinM :

Quebec / Ontrario:

We now port in usually 2 weeks. Last ~50 ports or so all ported in about 10 days.

We'll see! My port request for a 416 number went in Feb16, was processed on Feb18, so hopefully we'll see the port completed by Monday! I'll let you know.

Martin was correct! Port completed today!

Re: [Other] VOIP.MS

Sun, 01 Mar 2009 13:46:22 EDT

hockeynomad : Figured it could be the firewall,it pops up each time and I allow it but funny why it keeps popping up.

pinged the device and worked.

well can't do a update for now.

Re: [Other] VOIP.MS

Sun, 01 Mar 2009 13:16:00 EDT

Mango : That's curious. Are you able to ping the device? (Start>>Run>>cmd>>ping [IP address of device]).

Perhaps you have a firewall such as ZoneAlarm which is blocking the firmware upgrade utility?

Re: [Other] VOIP.MS

Sun, 01 Mar 2009 11:59:04 EDT

hockeynomad : I tried upgrading version 5.2.5 and 5.2.3 firmware of SPA 21202 but it couldn't connect to SPA device. :(

Re: [Other] VOIP.MS

Thu, 26 Feb 2009 14:55:27 EDT

bthornhill :

said by MartinM :

Quebec / Ontrario:

We now port in usually 2 weeks. Last ~50 ports or so all ported in about 10 days.

We'll see! My port request for a 416 number went in Feb16, was processed on Feb18, so hopefully we'll see the port completed by Monday! I'll let you know.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 15:09:49 EDT

hockeynomad : Echos can be managed by Echo cancellation card on your PC but at over $700 I will take a pass.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 14:05:42 EDT

Mango : Martin,

On behalf of the forum, let me say that we really appreciate this detailed explanation of the issues, especially your plans to ensure that they do not reoccur. It's very much appreciated.

Just try getting this kind of attention from Telus and you'll find out why I use VoIP :D

m.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 12:57:51 EDT

MartinM :

said by otty :

I had some serious quality problems this morning using the Toronto server. I quality on my end was fine, but the other end could barely hear me I was so choppy and garbled. I reattempted the call using both value and premium, g711 and g729, NY server and Toronto server. Same results.

Anyone else have these problems? Anyone have a possible explanation?

I do share my internet with neighbours, and judging from the lights on the modem they were actively using it, but I doubt using g729 I would have issues. Never did before.

If you are experiencing this issue on all calls, many servers and on both routes, the issue is probably due to latency, jitter or packet lost.

What I would recommend when this happen is to dial our Echo test, when the intructions are done playing, what you say is repeated back to you. This way you can verify the quality between you and the server, without involving any PSTN termination. IF you have issues at this stage, the problem is most likely on the internet connection or VoIP equipment. You can reach the echo test by dialing 4443.

If the problem happen only on specific server and/or numbers, send an email to support and we'll look at it and make appropriate changes if needed.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 12:41:27 EDT

MartinM : Porting Issues

Issue of Gkerr:

This has probably been the worst situation we’ve experienced so far and it made me very mad. This is a new distribution POP and we had a fiber cut due to an accident (Bell Nexxia). At the time this happened, our failover for this new incoming pop was still being installed in Montreal. Our vendor rushed the installation of this pop but it took some time. Our main system administrator was out of town at that time attending a VoIP course for future equipment upgrades. When the fiber cut was repaired, there was a detail that a technician forgot, and what should have been a routine and transparent change back to main incoming pop, caused your numbers to give a busy signal yet again. This is a very unfortunate series of events and although it affected a very small portion of our customer base (Around 10 numbers), because this is new equipment, we took the event very seriously and have had so far 2 meetings internally with our staff and a 2 hours conference with our vendor. We've taken measures to prevent this from happening in the future and we built a procedure code when a senior administrator is not available. I'm very sorry about the inconvenience this may have caused to your business. Although this probably doesn't compensate for the inconvenience you have experienced, I've waived the fee for your port orders. This has been discussed of course already with you, with Steve Poirier and feel free to continue your correspondence with him regarding this issue.

Quebec / Ontrario:

We now port in usually 2 weeks. Last ~50 ports or so all ported in about 10 days.

West Canada:

We've had numerous meetings with carrier and vendors for our rate centers there and have modified both our method to submit ports with improved LOA and verifications and especially, follow-ups, carrier and vendors will start processing faster. Our ports will now complete in 4 weeks maximum instead of the usual 4 to 8 weeks announced on our website,

USA:

Ports still completing normally. There are a few ports taking more time than usual but can happen under specific circumstances. Kbl, I would need your telephone number, although I see one port stuck from November so I will assume it's yours. It's being worked on by our carrier. We've had to re-submit LOA 2 times with some modifications due to Verizon being touchy regarding information provided on the LOA. It should complete soon, and we'll waive the fee for it, of course we'll keep you updated as we receive news. There are a few other ports being slow to complete, there are many causes to this. Rejections from losing carrier for address mismatch and various other things, when this happen we don't cancel the ports, we try to work around the issues so when this kind of situation happen, the port is delayed. Overwall, there are not many issues with US numbers being ported to VoIP.ms.

We're receiving a great quantity of porting requests since we first started. Our system is now inadequate to handle such a large quantity of requests and the rewrite of this portion of the website is almost done. I believe I explained earlier how it works. There will be no more fax. Only upload your invoice, signature and fill a much simpler form. We believe this will speedup the porting process by another week or two. ETA is Monday March 2.

Edit: Corrected a bad ETA Date

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 12:12:16 EDT

anon : Update to my DID problem with voip.ms - The DIDs were completely down for over 24 hours, apparently because of a fiber cut with their carrier. It was resolved temporarily, and Voip.ms assured me that the problem was solved. The DIDs then went down again yesterday for a couple of hours. I didn't know about it until I got an email from someone who was trying to call me, and when I called voip.ms they had no idea it was happening. It was solved again, but I still don't understand what went wrong. I talked to Steve Poirer on live chat, and he said they have fixed the processes so that there will be a working failover in the future, and will email me a summary of the new processes next week. I will post an update then.
In the meantime, I wouldn't recommend voip.ms for business-critical DID provision.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 11:45:19 EDT

Mango : What a great explanation Dan - Thanks!

I'd love to hear some updates on some of the issues that were posted on this thread so that the rest of us can attempt to avoid them ourselves if possible. gkerr, hopefully your DIDs are working by now. kbl, did you get that update?

m.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 10:15:52 EDT

engineerdan :

said by hockeynomad :

OK my only real concern has been callers on the other end of line as they must sometimes contend with echos, choppy voice (mine).

Normally I wouldn't mention this but, because I believe you're genuinely concerned...

If callers on the other end hear echos, the problem most likely has to do with your equipment. Most phones unintentionally "leak" some amount of RECEIVED audio back into the TRANSMITTED audio signal path. This is caused by regular, analog phones that simultaneously "listen" and "talk" on the same wires.

With plain, old telephone service, this leakage is not perceived as a problem because there is virtually no DELAY. Therefore, the caller hears themselves in real-time (no echo). However, because the Internet and VoIP has a certain latency (delay), previously unnoticed leakage becomes apparent to callers on the other end of the line as an echo.

Notably, in a VoIP environment, the "listen" and "talk" signals are transmitted separately from each other. For this reason, VoIP technology - by itself - cannot cause an echo.

Analog Phones contain a device called a HYBRID that reduces the leakage between the "listen" and "talk" signals. Sadly, not all HYBRIDS are created equally. Some phones have poor hybrids and, therefore, more perceived echo than others. If you have more than one phone, you may wish to experiment to see which phone has the least perceived echo (as judged by a distant volunteer).

IP Phones (Snom, Grandstream, etc.) maintain the separate "listen" and "talk" signals all the way to the handset. With such a phone, the only way a "listen" signal can leak into the "talk" path (causing echo) is through the air - by way of acoustic coupling from the handset earpiece to the handset mouthpiece. This sometimes happens when the volume is turned up too high on the earpiece and the user isn't holding the handset firmly to their ear.

With that, I hope we've helped to solve half your VoIP problems. I'll leave the choppy voice discussion to others.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 08:07:42 EDT

hockeynomad : OK my only real concern has been callers on the other end of line as they must sometimes contend with echos, choppy voice (mine). On my end I have to say the voice quality has been top notch.

There was an online tool someone posted that did a test of VOIP quality to a servor and back. Mine turned out the same there and back. That indicates no problem really at the time of test.

Re: [Other] VOIP.MS

Wed, 25 Feb 2009 00:11:49 EDT

PX Eliezer : Sounds like kbl is waiting 14 weeks.

Re: [Other] VOIP.MS

Tue, 24 Feb 2009 22:00:19 EDT

hockeynomad : 7 weeks? nothing's changed

Re: [Other] VOIP.MS

Tue, 24 Feb 2009 17:46:19 EDT

Mango : Just got my FOC and it's for TODAY!!

*paces around the room impatiently waiting* :D

Re: [Other] VOIP.MS

Mon, 23 Feb 2009 18:58:14 EDT

kbl : Unfortunately I'm also having problems with the porting process at Voip.MS as well.

Submitted early November 2008... nothing yet.
(Canadian number)

Support via Live Help has been much more helpful than via e-mail.

I should hopefully have an update tomorrow.

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 20:50:11 EDT

kaila :

said by otty :

...Just made a call to the same number from my University connection and the quality was better so it probably was bandwidth saturation. Thanks for the suggestions.

Reminds me of the adage that applies to an ITSP, a big company using voip, or a single user: Voip requirements are 80% networking and 20% telephony.

I've been using voip.ms for close to three months now primarily for outbound termination and haven't detected any issues with their Dallas server using IAX (premium routing). But as ambitious and aggressive as they are, I'm fully expecting a steady supply of growing pains to lengthen this thread.

Speaking of which, I know they haven't yet enabled the features that Martin posted about a couple of weeks ago, but when/if they do, it will really be pushing typical asterisk (or any PBX system) feature sets close to the edge (end user). Almost blurring (in my eyes) the line between ITSP and the typical hosted PBX provider. All this is a good thing, but they'll undoubtedly take a few shots to the chin getting all these changes into their system and really will need to keep the 80/20 rule in focus.

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 19:32:22 EDT

otty : Actually I forgot that I did a speed test earlier when I had problems:

Those speeds seem sufficient for voip using g729

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 18:34:34 EDT

otty :

said by Mango :

True, but it sounds to me like in Otty's case he's sharing a single connection with his neighbour using a router or switch. Apologies if I misread his post.

Yes you are right. Sharing the connection using a router. Just made a call to the same number from my University connection and the quality was better so it probably was bandwidth saturation. Thanks for the suggestions.

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 18:14:07 EDT

priller : Ah, didn't catch that part .. sorry

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 18:06:22 EDT

Mango : True, but it sounds to me like in Otty's case he's sharing a single connection with his neighbour using a router or switch. Apologies if I misread his post.

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 18:00:23 EDT

priller :

said by Mango :

It's true that g.729 uses very little bandwidth, however if your neighbour is running a torrent program or something similar that attempts to use all available bandwidth and then some, issues such as this will occur regardless of codec. A router with good QoS such as a WRT54GL with Tomato would likely solve the problem.

Any QoS you are doing on your router has no affect on network saturation by a neighbor's traffic. All you can do is give your VoIP traffic priority over whatever else you are generating.

No consumer ISP honors QoS markings set by an end user.

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 17:52:23 EDT

Mango :

said by gkerr :

My business has been completely unreachable for almost 24 hours - anyone calling one of our DIDs gets a fast busy signal.

MartinM, could we get some details on why this happened and why resolution took/is taking so long here in the forum please?

said by otty :

quality on my end was fine, but the other end could barely hear me I was so choppy and garbled. [...] I do share my internet with neighbours, and judging from the lights on the modem they were actively using it

I would bet money on this being an upstream bandwidth issue, since the issue occurs with multiple servers, and the audio issues are only in one direction. It's true that g.729 uses very little bandwidth, however if your neighbour is running a torrent program or something similar that attempts to use all available bandwidth and then some, issues such as this will occur regardless of codec. A router with good QoS such as a WRT54GL with Tomato would likely solve the problem.

m.

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 17:26:40 EDT

PX Eliezer : Right now, for me seems OK inbound and outbound, NY server, G.711U codec.

As I have posted previously, I have had some issues recently.

I hope that your problems improve, and I earnestly hope that gkerr has his business phone lines again!!

See also
»Re: voip.ms incoming call routing problems (fast busy tone)

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 17:26:35 EDT

otty : from what I understand g729 uses much less bandwidth and thus should be less likely to be affected by my neighbours using the internet while I make a call.

I never had this problem before. It's hard to track down if it is a problem with my internet connection or my ISP routing or with voip.ms

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 17:18:25 EDT

hockeynomad : I've had the same issues for a long time.

Audio configuration is set ar preferred codec g711u.

And in voip.ms account settings, at all codecs.

is g729 optimal?

Re: [Other] VOIP.MS

Sun, 22 Feb 2009 16:32:52 EDT

otty : I had some serious quality problems this morning using the Toronto server. I quality on my end was fine, but the other end could barely hear me I was so choppy and garbled. I reattempted the call using both value and premium, g711 and g729, NY server and Toronto server. Same results.

Anyone else have these problems? Anyone have a possible explanation?

I do share my internet with neighbours, and judging from the lights on the modem they were actively using it, but I doubt using g729 I would have issues. Never did before.

Re: [Other] VOIP.MS

Fri, 20 Feb 2009 13:43:15 EDT

hockeynomad : PM Martin ASAP.

Re: [Other] VOIP.MS

Fri, 20 Feb 2009 11:55:40 EDT

PX Eliezer :

said by gkerr :

I'm having a disastrous experience with Voip.ms right now (still ongoing; I will post updates when and if I get resolution). My business has been completely unreachable for almost 24 hours

Really sorry to hear that. :(

I've been unhappy with them lately for outbound calling, but the inbound DID service has been OK, even as of today.

Hope they get this fixed for you!

Please keep everyone posted.

----------------------------------

Martin M??

Re: [Other] VOIP.MS

Fri, 20 Feb 2009 11:38:48 EDT

anon : I'm having a disastrous experience with Voip.ms right now (still ongoing; I will post updates when and if I get resolution). My business has been completely unreachable for almost 24 hours - anyone calling one of our DIDs gets a fast busy signal. The DIDs don't work at all; even if I forward the DID to my mobile phone calls won't go through. So there is no failover, no voice mail... nothing. Essentially my business is closed.

I reported this to voip.ms live chat yesterday and was assured it would be corrected 'very soon'. As far as I can see, they didn't even submit a trouble ticket; I had to explain the whole situation this morning at about 8:30am, and as of 11:30am it is not only unresolved, they won't even give me an estimated time for resolution except to say it will be resolved "soon". Whatever that means.

Re: [Other] VOIP.MS

Wed, 18 Feb 2009 21:39:12 EDT

priller :

said by Mango :

said by priller :

If you are getting echo on the 4443 test number, you need to adjust FXS Input/Output gain, as previously mentioned.

Isn't 4443 supposed to echo? ;)

I was just laughing at myself!!! Doh!

I was assuming we were talking about unwanted, distorted feedback.

Re: [Other] VOIP.MS

Wed, 18 Feb 2009 21:37:57 EDT

otty : Ya my understanding is that an abrupt echo is what you want. A delayed echoe would mean that there would be a delay and echo effect in an actual call. Blurry probably isn't good though. Can't really imagine what blurry sound is though.

Re: [Other] VOIP.MS

Wed, 18 Feb 2009 21:33:27 EDT

Mango :

said by priller :

If you are getting echo on the 4443 test number, you need to adjust FXS Input/Output gain, as previously mentioned.

Isn't 4443 supposed to echo? ;)

Re: [Other] VOIP.MS

Wed, 18 Feb 2009 21:32:04 EDT

Mango : It looks like version 5.2.5 and 5.2.3 is available on »www.linksys.com/ . Installation is relatively simple; just run the .exe file from within the .zip file that you download.

If your RTP Packet Size is set to 0, my guess is it would use the default of 0.03. I suggest 0.02 or 0.01.

Re: [Other] VOIP.MS

Wed, 18 Feb 2009 21:28:10 EDT

priller :

said by hockeynomad :

P.S. also my RTP Packet Size is already set at 0 and getting echo.

0 ?!!

Make that 0.020

If you are getting echo on the 4443 test number, you need to adjust FXS Input/Output gain, as previously mentioned.

Re: [Other] VOIP.MS

Wed, 18 Feb 2009 21:21:29 EDT

hockeynomad : Found a firmware update to the 3.3.6 version of SPA2102.

»www.telefonujeme.cz/about2978.html

Its in Czech, though I don't expect you to translate, what are the step to update after I download this file?

P.S. also my RTP Packet Size is already set at 0 and getting echo.

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 12:42:44 EDT

Mango :

said by bthornhill :

Also, will the premium routing option improve call echo?

I have not experienced echo with the premium or value routes. Some suggestions:

I find that later versions of firmware seem to do a better job of handling echo on the Linksys devices I've tried. Additionally, echo can also be greatly reduced by adjusting FXS Input/Output gain or Handset Input/Output gain. I reduced one, then the other, in increments of 3.

One other thing you can try is setting your RTP Packet Size properly. The default on Linksys/Sipura devices is 0.03; try 0.02 or even 0.01. (Note that this will increase bandwidth usage slightly.)

Additionally, if your internet connection does not experience much jitter, (variations in ping times) set your jitter level to low. (If your internet connection does however experience lots of jitter, this will add to your problems. Change it back if things get worse.)

-m.

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 12:41:05 EDT

hockeynomad : 4443 echo returns a very abrupt and blurry echo.

using value.

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 11:09:21 EDT

otty : Here are the test numbers. These can be found under the account information tab.

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 11:06:01 EDT

N9MD :

said by bthornhill :

I thought I read somewhere that there was a number for echo test.

»Re: Phone numbers to test audio quality?

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 11:01:37 EDT

bthornhill : I thought I read somewhere that there was a number for echo test. Anybody know it?

Also, will the premium routing option improve call echo?

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 10:22:58 EDT

MartinM :

said by kaila :

Martin, any plans or chance Voip.ms will offer a wideband codec?

As long as you don't transcode down between servers (do you?), enabling the g722 codec would really put you guys over the top and differentiate your product from anything else available. Sure, I know it would require station-to-station g722 support, but it would be a mighty compelling feature to a growing number of us using hard/soft IP-phones.

We do not plan to offer this at this time. No carrier or vendors we have support it. From my personnal experience, you'll not gain any voice quality over g711u comparing to a standard PSTN line. And the conversation would have to be transcoded down to G711U.

I do believe voip.ms uses GSM between gateways anyway, which would kill all wideband benefits.

That's not true. The main reason why we recommend G711U and G729 and an optionnal G723, is to preserve the original codec from end user to carrier to prevent codec translation and loss of audio quality. By loss of audio quality I mean, for example you the customer using G711U and have your conversation transcoded to gsm or g729. That will not happen on any of our premium route. We also offer gsm as an alternative. This codec we'll transcode to g729 or ulaw depending on where you call and which route you are using.

I am completely unable to receive calls on my DID this morning, anyone experiencing this?

Service back...

I'm very sorry that you had issues but the service didn't go away and didn't go back this morning. I would suggest verifying your registration. If you still have issue with your DID number, please follow up via support via one of your tickets. Thank you

I just now got a confirmation (and a reply) from an email I sent four days ago. Hopefully the issue is solved for good this time.

Yes, the last batch of emails that were late was finished Saturday and this issue is now resolved for good.

edit: some typos

Re: [Other] VOIP.MS

Mon, 16 Feb 2009 00:05:03 EDT

Mango :

said by dpoliteski :

No confirmations are being sent out when you email support.

I just now got a confirmation (and a reply) from an email I sent four days ago. Hopefully the issue is solved for good this time.

Re: [Other] VOIP.MS

Sun, 15 Feb 2009 12:13:32 EDT

WildChild : Service back...

Re: [Other] VOIP.MS

Sun, 15 Feb 2009 11:20:28 EDT

WildChild : I am completely unable to receive calls on my DID this morning, anyone experiencing this?

Re: [Other] VOIP.MS

Sun, 15 Feb 2009 11:13:52 EDT

priller :

said by JasonOD :

said by kaila :

Martin, any plans or chance Voip.ms will offer a wideband codec?...

Add in the support nightmare this would cause (from customer issues to back end problems), and I doubt we'll see any provider using wideband for some time.

Uh ... Junction Networks OnSIP had been supporting G.722 for several months now.

Re: [Other] VOIP.MS

Sun, 15 Feb 2009 10:43:43 EDT

anon :

said by kaila :

Martin, any plans or chance Voip.ms will offer a wideband codec?...

Ha Ha! It would definitely 'diferentiate' voip.ms, but not in a positive way that you suggest. I do believe voip.ms uses GSM between gateways anyway, which would kill all wideband benefits. Ask Nitzan, VoipO Tim, or anybody else clued in to what providers have to deal with, and you'll see why wideband just isn't viable right now. Save your G.722 for between local extensions, or use asterisk to 'roll your own' and pass G.722 between remote points.

Yes handset makers (and some softphone makers) are now dumping G.722 enabled phones on the market, but they really have no idea what it it's going to take to support it. For starters, Asterisk 1.4.x doesn't play well with G.722. Asterisk 1.6.x does, but no provider would dare use it in a production environment. Add in the support nightmare this would cause (from customer issues to back end problems), and I doubt we'll see any provider using wideband for some time.

Re: [Other] VOIP.MS

Sun, 15 Feb 2009 09:49:57 EDT

kaila : Martin, any plans or chance Voip.ms will offer a wideband codec?

As long as you don't transcode down between servers (do you?), enabling the g722 codec would really put you guys over the top and differentiate your product from anything else available. Sure, I know it would require station-to-station g722 support, but it would be a mighty compelling feature to a growing number of us using hard/soft IP-phones.

Re: [Other] VOIP.MS

Sat, 14 Feb 2009 12:44:14 EDT

dpoliteski : I think there is an issue currently with the support system. No confirmations are being sent out when you email support. So I don't think they are seeing the issues, then not being able to reply to them. The original issue might be fixed with ticketing, but now might be a new one. Martin do you see this? Thx.

Re: [Other] VOIP.MS

Fri, 13 Feb 2009 13:11:15 EDT

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 16:26:19 EDT

Mango : SWEET!!! You absolutely ROCK! :)

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 16:21:40 EDT

MartinM : I see what you mean. I've recommended to the dev team for the CNAM query to not take place if there's already a CallerID Name passed to us. This should go in production at the same time the "caching" system takes place in a couple of week. Thank you for your excellent recommendation.

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 14:10:43 EDT

Mango : CNAM turned off for Canadian DIDs works for Cable/Cell/VoIP phones but not POTS phones. CNAM turned on works for POTS phones. I'm just picky and want the best of both worlds :D This is obviously not a huge concern, but if it were one day possible to do a CNAM query only if SS7/IAM data was not passed, that would make an already-great service even better.

Great to hear you're deploying some new servers. It's so great that I can switch servers in a few minutes if my ISP decides to have a routing issue.

Ironically, my colo provider is in Michican and I've actually never used a Vancouver datacentre. I do know of two local VoIP providers, DigitalVoice (not recommended) and NetVoice (recommended, but expensive for residential users) that use GroupTelecom. (Curiously, TekSavvy routes western connections to GT through Seattle but this only adds a negligible 6ms.)

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 13:36:27 EDT

MartinM : I believe you can keep CNAM turned off for your Canadian DIDs. Let me know if you receive the callerid that is passed to the numbers with CNAM turned off. If it doesn't work to your liking, let me know we'll arrange something.

While we're talking about Vancouver, we want to deploy 2 new servers in the west. Specifically, Vancouver or very close to it and Calgary, or surroundings close to it. Let me know if you have any good data center recommendation in that area.

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 12:55:44 EDT

Mango :

said by MartinM :

We've had some issue with the support in the last 2 weeks that made many tickets get set to the wrong status. We've catched up with the issues and answered the majority of late requests. For that, I apolgize.

Great to hear this is now sorted - I was starting to get worried :)

Martin, any idea about my Caller ID Name question? In case you missed it, will there ever be the option to do a CNAM query only if SS7/IAM data is not passed? Here in Vancouver, all POTS numbers are in a CNAM database and their lines do not pass SS7/IAM data whereas Cable/Cell/VoIP phones are the reverse. So with the current system, I can choose to see Caller ID Name on one half of my calls or the other half of my calls, but not both.

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 12:49:01 EDT

MartinM : I'm just adding the reason why your DID not answer and the voicemail picked up.

Of course if you have voicemail enabled on our side, and your system is not reachable, voicemail on our side will kick-in.
This is working as intended.

Please feel free to continue this discussion in PM or simply reply to your ticket for further assistance.

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 12:14:11 EDT

hockeynomad : That's not what Stef at customer support said. She indicated maybe my own answer device wasn't working and it went to the Voip.ms messaging. If I didn't like it, I could disable the the voip.ms from my profile. That was the answer.

"Dear Customer,

Our our technicians review your problem and didnt find any problem on our side, maybe your voicemail device didnt work and the voicemail that you setup on your DID, If you dont want that our voice mail comes up, you should disable it, so your answer machine will always come up

Sincerely,

Stef ""

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 11:56:14 EDT

MartinM :

said by dpoliteski :

I ended up cancelling Acanac as I couldn't get the reliability worked out. I am trying voip.ms now for 2 weeks so far. Generally call quality not bad, recently outbound calls not sounding that great in both premium and value. Also noticing that outbound callerid not always working properly either. Might just have to keep my local Telus line after all. Support at voip.ms is not good at all as they take forever tog et back to you. If I can get things worked out then may stay with them, otherwise VOIP again out the window.

We've had some issue with the support in the last 2 weeks that made many tickets get set to the wrong status. We've catched up with the issues and answered the majority of late requests. For that, I apolgize.

It sound that your issue may not be related to the providers after all. I would recommend dialing "4443" for an echo test, and if you have quality issues doing this, your connection or equipment may not be appropriate for VoIP. If, at the contrary, your echo test is fine, send me an email and we'll try to find a solution to your network issues.

Re: [Other] VOIP.MS

Thu, 12 Feb 2009 11:52:50 EDT

MartinM :

said by hockeynomad :

OK had an issue last night as normally my phone did not ring and went directly to voip.ms voicemail.

My setup entails on the 4th ring it goes to my phone answering machine and it only goes to voip.ms system voicemail if I am on the phone or if DSL/VOIP are down.

At the time i was streaming a video and downloading/uploading files in the background.

I should contact voip chat support, but my inclination is its a waste of keystrokes.

If DID number goes directly to your voicemail it's because the registration contact the server had at this time was unreachable and the call failed to reach your device properly.

You would have received that answer by support as well, and quicky than on the forum.

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 21:07:54 EDT

PX Eliezer :

said by bthornhill :

CALL PARK/PICKUP:

Does anyone know if I can configure call park/pickup on voip.ms (no PBX). If so, what are the codes?

I've only heard of this as a feature of Asterisk or other PBX systems, and some specific ATA's or IP phones (code *38)

I have not seen this as a feature of any Voip provider, and I've looked at many.

You may want to post this as a new topic, not mentioning the specific provider, just because more people may see it and maybe someone else will know of a provider that supports it.

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 20:12:27 EDT

bthornhill : CALL PARK/PICKUP:

Does anyone know if I can configure call park/pickup on voip.ms (no PBX). If so, what are the codes?

TIA,
B

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 13:13:12 EDT

hockeynomad : OK had an issue last night as normally my phone did not ring and went directly to voip.ms voicemail.

My setup entails on the 4th ring it goes to my phone answering machine and it only goes to voip.ms system voicemail if I am on the phone or if DSL/VOIP are down.

At the time i was streaming a video and downloading/uploading files in the background.

I should contact voip chat support, but my inclination is its a waste of keystrokes.

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 11:34:47 EDT

WildChild : I have absolutely no problem with outgoing calls. It seems the problems may be related to routing to the DID as when I try to call it and it fails, there is no log of the failed call on the CDR in voip.ms web interface (no unanswered, busy or failed log entry). Also, yesterday and early this morning I was testing with my Videotron line and about 4 times out of 5, I got the fast busy tone. With Bell at work, I get a busy tone about 1 time out of 20-25 tries. As I said, every time I get the busy tone, nothing get logged in the CDR. I already contacted the Online Chat of voip.ms. I'm waiting for a follow up from them.

said by Yippz Voip :

As Priller said, remove all the port forwarding, open port stuff etc etc from your router -- you don't need to do ANYTHING to your router to make voip.ms work inbound and outbound.

Then, log into your Control Panel on Voip.MS and under SUPPORT look for CONFIGURATION SAMPLES. Goto the LINKSYS section and you will see about 3-4 things that you need to enter on your adapter do to make your adapter work. Also, since you're in Quebec, I assume you're using the Montreal server.

If all else fails, reset both your router and Linksys adapter to FACTORY RESET to get rid of the changes you made that are causing the problem and start from the beginning using the config samples from voip.ms

Lastly, if all setting are correct and you are having inbound ONLY problems from a DID, it could be the routing of the DID. I had an occurance where one of my inbound numbers rang "OUT OF SERVICE" -- had nothing to do with my equipment, it was on their end. A simple chat on their website and the problem was resolved. It had something to do with one of their providers...

So if all else fails, Voip.MS has LIVE HELP which is available all day and well into the late evening hours for further assistance.

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 11:18:42 EDT

Yippz Voip : As Priller said, remove all the port forwarding, open port stuff etc etc from your router -- you don't need to do ANYTHING to your router to make voip.ms work inbound and outbound.

Then, log into your Control Panel on Voip.MS and under SUPPORT look for CONFIGURATION SAMPLES. Goto the LINKSYS section and you will see about 3-4 things that you need to enter on your adapter do to make your adapter work. Also, since you're in Quebec, I assume you're using the Montreal server.

If all else fails, reset both your router and Linksys adapter to FACTORY RESET to get rid of the changes you made that are causing the problem and start from the beginning using the config samples from voip.ms

Lastly, if all setting are correct and you are having inbound ONLY problems from a DID, it could be the routing of the DID. I had an occurance where one of my inbound numbers rang "OUT OF SERVICE" -- had nothing to do with my equipment, it was on their end. A simple chat on their website and the problem was resolved. It had something to do with one of their providers...

So if all else fails, Voip.MS has LIVE HELP which is available all day and well into the late evening hours for further assistance.

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 08:22:25 EDT

WildChild : Why removing the port forwarding would resolve the problem? I also found something about enabling Keep Alive on the ATA so the NAT router keeps the connection registred. I configured a Keep Alive of 60 secs on the ATA and a UDP Keep Alive of 300 secs on the router so the ATA has enough time to ping voip.ms without the router dropping the connection. Now I'm at work and it seems I can't get this fast busy tone from here... Only from my home Videotron's telephone line. Can it be related to Videotron?

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 08:08:35 EDT

priller :

said by WildChild :

I can sometime reach it but more often, I get a fast busy tone. What can cause this? I'm behind a router and I forwarded the 5060 port TCP/UDP and the 16384-16482 ports (found on the SIP tab of the adapter) UDP to the adapter.

Remove the port forwarding you configured and see if that resolves the problem.

Re: [Other] VOIP.MS

Wed, 11 Feb 2009 07:58:43 EDT

hockeynomad : Seems through this thread one has more influence as opposed to the chat support, as Martin appears to have an ear to certain concerns.

Re: [Other] VOIP.MS

Tue, 10 Feb 2009 22:34:13 EDT

WildChild : Hello, I received a Linksys SPA2102 to try voip.ms today. Everything seems fine, I can make call, I can receive call, audio quality seems fine to me, the DID was setuped quite quickly today. The problem I have is that when I try to call my DID from my current landline, I can sometime reach it but more often, I get a fast busy tone. What can cause this? I'm behind a router and I forwarded the 5060 port TCP/UDP and the 16384-16482 ports (found on the SIP tab of the adapter) UDP to the adapter.

Thanks

Re: [Other] VOIP.MS

Tue, 10 Feb 2009 21:34:33 EDT

PX Eliezer :

said by dpoliteski :

I am trying voip.ms now for 2 weeks so far. Generally call quality not bad, recently outbound calls not sounding that great in both premium and value. Also noticing that outbound callerid not always working properly either. Might just have to keep my local Telus line after all. Support at voip.ms is not good at all as they take forever to get back to you. If I can get things worked out then may stay with them, otherwise VOIP again out the window.

Thanks for letting us know.

Would be interested to see the company's response!

Re: [Other] VOIP.MS

Tue, 10 Feb 2009 20:50:04 EDT

dpoliteski : I ended up cancelling Acanac as I couldn't get the reliability worked out. I am trying voip.ms now for 2 weeks so far. Generally call quality not bad, recently outbound calls not sounding that great in both premium and value. Also noticing that outbound callerid not always working properly either. Might just have to keep my local Telus line after all. Support at voip.ms is not good at all as they take forever tog et back to you. If I can get things worked out then may stay with them, otherwise VOIP again out the window.

Re: [Other] VOIP.MS

Tue, 10 Feb 2009 20:26:38 EDT

RALMAR : Don't use the Acanac VOIP as a main line because sometimes you are going to fail to open the building door (Canada Post/Purolator is going to fail to contact you). You are going to pay 10$ + YOU are going to spend more than one hour per month to write emails when the phone is not working.(you don't have this issue in the first trial month) + the time spent to fix the issues(because the phone was down and you need to go to Canada Post/Purolator or somebody else).
In the end ACANAC is too expensive for me.

Good luck...

Re: [Other] VOIP.MS

Tue, 10 Feb 2009 11:01:15 EDT

dcdeadbeat : The Aastra phones should work fine behind a router. However, if you have not purchase the 9xxx phone, I would recommend that newer Aastra 6730i. It is not that much more expensive and will support G.722 wideband calling. Voip.ms does not currently support provide support for G.722, but if you are connecting multiple phones using their sub-account extensions, then you should be able to call each phone using G.722.

One problem you may find is that the more wideband calls you make, the more you realize how bad regular phone calls over the PSTN sound.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 21:04:50 EDT

priller : Well, I have a Polycom IP 550 registered directly to voip.ms (behind NAT). Just plug the thing in and you should be fine. Ignore all the port forwarding and DMZ nonsense you read.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 20:48:56 EDT

Mango :

said by bthornhill :

I am planning to use Aastra 9112i terminals

*drools* WANT!!!!!!

said by bthornhill :

These terminals do not support PPPoE network config, just DHCP or manual config. That means that they will have private IP addresses instead of public ones. Will this create a problem, or will NAT work OK?

I have devices connected behind NAT, using Shaw Cable and TekSavvy DSL. I did turn on the NAT setting in VoIP.ms' control panel and also NAT Keep-Alive in the adapter's settings.

No port forwarding, no DMZ, no special router config at all. Everything works just fine.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 20:06:49 EDT

bthornhill : I am planning to use Aastra 9112i terminals on a voip.ms account in my home. I have Bell DSL.

These terminals do not support PPPoE network config, just DHCP or manual config. That means that they will have private IP addresses instead of public ones. Will this create a problem, or will NAT work OK?

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 19:01:49 EDT

PX Eliezer :

said by priller :

said by hockeynomad :

I am a Linksys ATA 2102 and after speaking with Linksys support was told this device does not allow you to configure any changes for this.

Then the person you talked to is an idiot.

As Priller said, look under the Line tab (or Line1/Line2) for the DTMF settings.

I just wanted to add, Voip.MS is unusual in that there is also a DTMF setting on their control panel (after you log in to the Voip.MS website). It would probably be best to match the DTMF setting on the ATA with the DTMF setting on the Voip.MS website control panel.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 18:59:49 EDT

Mango :

said by PX Eliezer :

Regarding ports, though, they have said that it can take a few weeks.

It's not a port; I just asked them if they could get a particular DID for me. Their carrier already has the NPA/NXX. I guess it's not a usual request for them.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 18:56:50 EDT

priller :
I ported from Vonage to voip.ms and it took exactly 3 weeks.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 18:55:30 EDT

PX Eliezer :

said by Mango :

I really wish they'd improve the response time and overall quality of their technical support. I bet they'd hit the top if they did that.

That is good advice. I hope that Voip.MS is smart enough to act on it.

Regarding ports, though, they have said that it can take a few weeks.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 18:26:25 EDT

hockeynomad : DTMF TX Method was always set at Auto.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 15:11:01 EDT

Mango :

said by Seeley :

Voip.ms fell a bit on the charts -whether it means anything or not. »/gbu

I really wish they'd improve the response time and overall quality of their technical support. I bet they'd hit the top if they did that.

I've had a ticket about assigning a DID open since January 7. A few times a week I email in and ask for an update. They say they're working on it. A shame, because as far as call quality and reliability goes, they couldn't be better (as far as my experiences go.)

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 13:01:17 EDT

priller :

said by hockeynomad :

I am a Linksys ATA 2102 and after speaking with Linksys support was told this device does not allow you to configure any changes for this.

Then the person you talked to is an idiot.

Take a look at the "DTMF TX Method" under the Line configuration ... »ui.linksys.com/files/SIPURA/SPA-···oice.htm

In any event, "AUTO" works fine with voip.ms. If you still have problems, use the voip.ms Premium route instead of Value.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 12:16:09 EDT

hockeynomad : Had problems with touch dial capability after a call is made out and someone mentioned adjsuting the DTMF mode.

I am a Linksys ATA 2102 and after speaking with Linkss support was told this device does not allow you to configure any changes for this.

Re: [Other] VOIP.MS

Mon, 09 Feb 2009 11:04:34 EDT

Seeley : Voip.ms fell a bit on the charts -whether it means anything or not. »/gbu

Re: [Other] VOIP.MS

Sun, 08 Feb 2009 14:15:56 EDT

hockeynomad : Despite a few issues, I am sticking with voip.ms. Call quality on my end is great, and they seem to try to address some issues, though #1 on list is customer service responsiveness.

In the long haul, they should be OK and their rates are fine.

Callcentric is the only one I'd consider but I can port my number, yet.

Re: [Other] VOIP.MS

Sat, 07 Feb 2009 19:12:26 EDT

PX Eliezer :

said by Seeley :

I saw that. I have no idea what my modem's(Zoom 5241) ip address is. It's not impotant. I was just wondering if that has anything to do with my dropped calls.

Maybe. But my comment was a direct response to a question posted by wgraz, not by you:

said by wgraz :

How does one find out the SNR on the cable modem?

So I was answering his question.

Re: [Other] VOIP.MS

Sat, 07 Feb 2009 18:57:59 EDT

Seeley : I saw that. I have no idea what my modem's(Zoom 5241) ip address is. It's not impotant. I was just wondering if that has anything to do with my dropped calls.

Re: [Other] VOIP.MS

Sat, 07 Feb 2009 18:35:49 EDT

PX Eliezer :

said by wgraz :

How does one find out the SNR on the cable modem?

»www.speedguide.net/read_articles.php?id=1197

Re: [Other] VOIP.MS

Sat, 07 Feb 2009 17:54:22 EDT

anon : How does one find out the SNR on the cable modem?

Re: [Other] VOIP.MS

Sat, 07 Feb 2009 09:26:34 EDT

hockeynomad : Martin said:

CallerID Blocking: You will be able to block Anonymous Calls and Specific CallerID Numbers (Eta 1-2 weeks)

----------------------------------------------------------

Martin, can you add 8** call blocking as well?

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 16:17:25 EDT

PX Eliezer : Thanks for the followup. Glad it worked out.

In my situation, it seems to be specific to the one VoSP, and not relate to modem, router, ATA, or ISP.

But I have moved on. In Voip, as in love, sometimes things are just not meant to be.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 14:50:28 EDT

priller :

said by PX Eliezer :

In those posts, priller and wgraz both report dropped calls and other problems.

For a while there almost every call was being dropped. I now suspect that Comcast was the culprit .... the SNR on the cable modem was way too low ... Comcast made a fix down the street and I'm good now. No more dropped calls.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 14:45:57 EDT

Mango :

said by PX Eliezer :

»Re: [Other] VOIP.MS
In those posts, priller and wgraz both report dropped calls and other problems.

For whatever reason, it just now occurred to me that these are very similar symptoms to the dropped call/router issue I was having. My router's log showed SYN FLOOD ATTACK DETECT. For whatever reason, this did not happen with my previous provider.

I can't remember if I mentioned this to you guys or not, but in any case, I swapped the router (D-Link WBR-2310) for a better one I had lying around anyway, and the problem has vanished.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 14:30:05 EDT

Mango :

said by MartinM :

I will provide some updates regarding Incoming features for VoIP.ms, as requested by many customers.

Awesome :) Thank you, MartinM.

said by MartinM :

A big change regarding CNAM Queries: [...] CNAM Directory / Speed DIal:

This sounds great. Will there be the option to do a CNAM query only if SS7/IAM data is not passed? Here in Vancouver, all POTS numbers are in a CNAM database and their lines do not pass SS7/IAM data whereas Cable/Cell/VoIP phones are the reverse. So with the current system, I only see Caller ID on about half of my calls.

said by MartinM :

We'll add custom fail over conditions soon, that will let you program different routes according to the 3 following conditions: No Answer, Line Busy, Device or PBX Unreachable

SUH-WEET!!! I'm looking forward to this one.

Thank you very much for your presence in this forum, Martin - this is one of the main reasons I use VoIP.ms, besides of course the superb call quality.

-Mango

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 13:39:01 EDT

hockeynomad : I had outgoing calls that dropped in that same time period, after a few seconds and the other party could not hear me.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 13:20:48 EDT

PX Eliezer : Thanks for your response, much appreciated.

One main issue is calls that never connect. These don't show up on the CDR in my experience.

I have seen various recent posts reporting similar issues.

See, for example:

»Re: [Other] VOIP.MS

In those posts, priller and wgraz both report dropped calls and other problems.

There are others as well, I believe.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 13:18:11 EDT

MartinM : I will provide some updates regarding Incoming features for VoIP.ms, as requested by many customers.

Calling Queues, IVR (Digital Receptionist) and Recordings are currently in testing phase. This mean it's ready for a launch in the next few days. (Eta 1 week at most for launch)

A big change regarding CNAM Queries: We're reworking the system to cache your queries on a 24 hours basis. That mean that if "John Smith" Calls you 5 times in the day, you will be charged for only 1 Query. (Eta 14 days)

CNAM Directory / Speed DIal: You will be able to create entries with specific number. When you will receive an incoming call, if the CallerID Number matches one of the entries you have programmed, no Querie will be made to the CNAM/LIDB repository. It will instead display the name you have programmed for that entry. Eta: 15-25 Days

CallerID Blocking: You will be able to block Anonymous Calls and Specific CallerID Numbers (Eta 1-2 weeks)

Features codes: Most of the features that are accessible via the web interface, will have feature codes added to them. This will allow you to enable/disable features by dialing specific codes without havingto log in to the interface. I do not have a list of the specific features that will be added via this, but most standard features will be added (Last Caller, block last caller, Activate forwarding etc). I will provide the list when we are done with the planning. (Eta: 4-6 weeks)

Inbound DID Fail over according to condition: At the moment, fail over, if enabled, is activated on the following conditions: Busy or Unreachable. We'll add custom fail over conditions soon, that will let you program different routes according to the 3 following conditions: No Answer, Line Busy, Device or PBX Unreachable

I would like to thank everyone for their suggestions. Please keep them coming, we appreciate every suggestions and try to integrate the majority of them.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 13:06:53 EDT

MartinM :

said by PX Eliezer :

said by hockeynomad :

Can't wait for Caller-ID Block and Anonynous-caller block.

I admit that I endorsed Voip.MS in the past. But I am honest enough to say that I can't wait to use up my funds with them and move on.

cf:
»Review of voip.ms by PX Eliezer

I'm rally sorry to hear that you are not happy anymore with the quality of service. While you may not be experiencing the issues you are describing with other providers, our customers have not reported any specific issues either and our termination minutes keep increasing. I'm not saying that the problem is specifically on your side, but it doesn't mean it's a specific voip.ms problem either.

If you can send me some CDR entries via PM, I will gladly investigate this issue for you.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 11:16:22 EDT

hockeynomad : OK, can I use Callcentric strictly for termination calls, and voip.ms for incoming and 911?

All I need to do is set up a 3nd ATA device? I have a Mediatrix for a second device.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 11:08:43 EDT

PX Eliezer :

said by hockeynomad :

Can't wait for Caller-ID Block and Anonynous-caller block.

I admit that I endorsed Voip.MS in the past. But I am honest enough to say that I can't wait to use up my funds with them and move on.

cf:
»Review of voip.ms by PX Eliezer

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 11:02:34 EDT

hockeynomad : On caller ID on some calls I get "POTS PHONE" with the number. Is that because it is an unlisted number?

A nice feature for incoming calls is on my phone display it will show "Blocked" but looking in my account on-line, the number is displayed.

Can't wait for Caller-ID Block and Anonynous-caller block.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 04:40:53 EDT

AndrewZ :

said by dpoliteski :

If I want to also use my N95 to connect to make calls I guess I need to disconnect my ATA everytime I connect my N95?

Don't guess, just configure and use it. There is no need to disconnect other SIP-devices you may have.

Re: [Other] VOIP.MS

Fri, 06 Feb 2009 03:38:30 EDT

dpoliteski : Hi, I am currently connected to VOIP.MS using my Linksys ATA. If I want to also use my N95 to connect to make calls I guess I need to disconnect my ATA everytime I connect my N95? Any way people might know to have both devices connected at the same time?

Thanks,
Dan.

Re: [Other] VOIP.MS

Thu, 05 Feb 2009 08:50:20 EDT

PX Eliezer :

said by priller :

Well, I use the SPA9000 for my home system, as do other people I know.

Just curious, do you have IP phones connected to it, or using regular phones with additional adapters, or something else?

Re: [Other] VOIP.MS

Thu, 05 Feb 2009 08:07:59 EDT

priller : Well, I use the SPA9000 for my home system, as do other people I know. Guess it depends on your needs and the geek factor. For $279, it's a very slick system.

The question was how to use multiple providers. It's a very viable solution that anybody can take it or leave it.

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 22:37:02 EDT

PX Eliezer :

said by priller :

Another option is a small IP PBX, like a Linksys SPA9000, that allows you to have multiple providers. Very easy to have one provider for inbound, another for outbound. From a user perspective, it's totally transparent.

Good choice for some folks, but isn't it overkill for Seeley who just wants 1 phone and 2 Voip services?

The SPA9000 costs almost $ 300, and the phones connected to it (allows up to 4, optionally 16) have to be IP compatible (or connected through additional adapters). Good for a business, maybe not good for a basement.

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 22:36:11 EDT

otty : I think voxalot is probably the cheapest easiest solution, but just so you know many IP phones including the one I have (Nokia E51) allow you to be simultaneously registered with multiple SIP addresses.

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 22:27:20 EDT

priller : Another option is a small IP PBX, like a Linksys SPA9000, that allows you to have multiple providers. Very easy to have one provider for inbound, another for outbound. From a user perspective, it's totally transparent.

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 21:23:31 EDT

PX Eliezer : May be easiest to use a free service like Voxalot which lets you use multiple Voip carriers.

You'd set up the various accounts with Voxalot, then set up Voxalot on your ATA.

There are other services too, as I'm sure other folks will be posting....

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 21:18:26 EDT

Seeley : I'm thinking of using VOIP.MS just for Termination and Callcentric for a DID(and some Termination to CC members). Is there a method of using both carriers (each on one port of an ATA) and using one phone successfully. If I use a simple splitter, incoming calls shouldn't be an issue but how would I set it up so I can indicate which service to dial out on? -without buying a 2-line phone and without physically swapping the wires? Thanks.

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 21:16:56 EDT

PX Eliezer : Again, this is a DTMF issue. Cf that topic in these forums. But DTMF affects most Voip services....

There is a DTMF settings selector on the Voip.MS customer portal (webpage) under Settings---after you log in.

There is also almost certainly a DTMF setting on your ATA.

You may need to see which setting works best.

And make sure that the Voip.MS customer portal DTMF setting, and the ATA DTMF setting, are the SAME in each case.

I have had more of these issues with Voip.MS than with other carriers.

Because of outbound calling problems in general, I am shifting more usage from Voip.MS to CallCentric (even though costs more) and to CallWithUs. Have just updated my review of Voip.MS to reflect this.

Re: [Other] VOIP.MS

Wed, 04 Feb 2009 20:29:47 EDT

hockeynomad : From time to time I cannot use the touch-tone prompts; for instance when dialing out, pressing "1" for english, does not work.

Anyone else experience this?

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 19:13:20 EDT

burgerwars :

said by priller :

said by otty :

These are all unanswered test calls.

Yes I get charged for unanswered calls as well.

otty --- It looks like your calls are incoming to your DID and then you have it forwarded to your cell .... failover testing?

Yes, these calls will be charged since voip.ms has to turn that call around and place it back out the PSTN.

If you have a DID without failover/forwarding, there will not be a charge if you don't answer. As in the example I included earlier.

Also too me it looks like you're forwarding, so you're getting billed for both the incoming and outgoing part. I've tried this myself, and if the forwarded part of the call isn't answered, is busy, or is forwarded to a disconnected number I usually wasn't charged. I think it's called "false supervision" when a call is (or is not) answered on the other end but the opposite is detected by their software. So if that's the case, you could get charged. I've also found sometimes if I hang up on a call to my voip.ms number, my number will continue to ring for eternity. Maybe it does that on a forward.

One way around that, as mentioned, is set no sort of failover forwarding (or have voicemail pickup), if the phone isn't answered or your adapter is off. I find if I shut everything down this way, the caller just gets a busy signal. No charge there.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 17:35:11 EDT

otty : I think you are right. I believe I was using a ring group but I guess that's the same deal.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 17:15:45 EDT

priller :

said by otty :

These are all unanswered test calls.

Yes I get charged for unanswered calls as well.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 17:00:58 EDT

anon : Check unlimitel.ca I think they're in CDN Dollars.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 16:41:53 EDT

otty :

said by RockyBB :

they appear to be billing in fractions of cents per call ... four decimal place rounding.......4 decimal place rounding is a true gift to their customers.

Looks like 6 decimal places. At some point they do round up the total to the nearest cent (I read that somewhere but can't find it now). One of the reasons I really like these guys :D
--
Sasktel HighSpeed Extreme , Vista, AMD Turion 64 X2 TL-60 Dual-Core @ 2.0GHz; 2GB RAM

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 16:32:38 EDT

RockyBB : I can't speak to the billing of unanswered calls, and I can't see the bottom of your bill, but they appear to be billing in fractions of cents per call ... four decimal place rounding. In the traditional telecom US long distance business, it's unheard of for a carrier to bill in 4 decimal places ... essentially there's no rounding up to the highest penny per call, which is where carriers make a ton of money. Sure, they are billing in what appears to be 6 second increments (or tenths of minutes), but that 4 decimal place rounding is a true gift to their customers.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 16:14:54 EDT

otty : Ya no one has any Saskatchewan DIDs as it is all still publicly owned by Sasktel and I guess they don't cooperate with anyone.

voip.ms is still the best for me because I don't even use $10 per month and voip.ms has the best rate for pay as you go DID's at $1.99(USD :mad: ) per month.

Hmm. I don't trust acanac very much from what I've read on the Canadian Broadband forums but that does look like a good deal. Maybe if I notice I'm above about $8/month (US) it would be worth a try at $10 CDN.

EDIT: Watch out...I think thats a 1 year contract and I hear they aren't very forgiving on that.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 16:12:02 EDT

dpoliteski : There is Acanac which I am trying now. They are trying to work out some call quality issues and reliability for me. They are $10 a month including everything. If they can't get quality worked out $10 is meaningless as it wouldn't be usable. But if they can, it is a much better deal than voip.ms especially for those in Canada. I don't think I noticed any Sask DID numbers though for Acanac, for VOIP.ms either. I noticed you from Saskatoon.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 16:07:00 EDT

otty : These are all unanswered test calls.

Yes I get charged for unanswered calls as well.

I am in Canada and am not pleased about paying in US dolars either. I have yet to find any voip provider that charges in Canadian dollers though...all the businesses are cross-border and thus use US dollars as probably the majority of their customers are US based.

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 15:49:42 EDT

priller :

said by dpoliteski :

I am trying VOIP.MS out now, and am noticing their incoming rates are expensive as you get charged for people calling even if you don't answer.

Double check that. I do not have charges for unanswered calls. It shows the rate, but 0 minutes, $0.00 billed (see capture above).

Re: [Other] VOIP.MS

Mon, 02 Feb 2009 15:34:43 EDT

dpoliteski : I am trying VOIP.MS out now, and am noticing their incoming rates are expensive as you get charged for people calling even if you don't answer.

Other thing for any non US customer, the exchange rate can kill any deal that you thought you might get. I am in Canada, and basically in my mind I need to add 25% to the prices I see per minute.. Makes it not all that great a deal.

My 2 cents.

Re: [Other] VOIP.MS

Sun, 01 Feb 2009 23:41:10 EDT

Seeley : Because some Caller ID names are unknown due to incoming cellphone calls etc, Voip.MS NEEDS to make Contact Lists available.(like callcentric) If they did that, I would decide to go with Voip.ms solely.

Also with the dedicated Voip thread, can a moderator copy this thread to there? I think it would be a great idea. Thanks

Re: [Other] VOIP.MS

Fri, 30 Jan 2009 13:52:30 EDT

Guiness :

said by priller :

My plea to all: It's OK to start a new topic when you have a question about voip.ms. It doesn't all have to go in a single mega thread, really. This has become counterproductive.

Mods: Any chance we could have this one locked?

An even better idea would be to use voip.ms 's dedicated forum?

»voip.ms

Re: [Other] VOIP.MS Caller ID

Fri, 30 Jan 2009 11:45:50 EDT

otty : Yup turn the feature off it works. However you won't get a name but you do get the number that's calling. Interesting it seems he was getting the location? My parents are getting that on their POTS line now too. Never heard of it...If you know most of the area codes that is kind of useless anyways.

Re: [Other] VOIP.MS Caller ID

Fri, 30 Jan 2009 09:03:05 EDT

PX Eliezer : Well, it's not locked yet.

So for anyone reading this---

The FIX for the Caller ID Name problem, IF you are in Canada, is to turn the feature OFF.

See:

»Caller ID Question

Re: [Other] VOIP.MS

Fri, 30 Jan 2009 08:23:39 EDT

priller : To address the *97 question, follow this same thread from this point .. »Re: [Other] VOIP.MS

There is nothing you can do to further enhance the CID name being delivered.

-------------------

My plea to all: It's OK to start a new topic when you have a question about voip.ms. It doesn't all have to go in a single mega thread, really. This has become counterproductive.

Mods: Any chance we could have this one locked?

Re: [Other] VOIP.MS

Fri, 30 Jan 2009 03:06:27 EDT

dpoliteski : Hi all,

I signed up for VOIP.MS yesterday and am testing out the features and call quality. So far so good pretty much.

Couple questions for any of you VOIP.MS experts...

1. Inbound caller ID.. I have it set on the customer portal for the DID to enable caller id with the small fee listed. I get a name when a call comes in but it either says Cell Phone, or British Columbia for example. It doesn't show the actual name of the user from telco database. Is this a known issue or would I need to set something else up? I am using Linksys/Sipura SPA-2102 as my ATA device.

2. To check voicemail, the say use *97 + mailbox number. I am not able to dial that (maybe ATA Setting). I can *98, and then am asked for mailbox number and password. Is there a way to have *98 actually enter the required mailbox number to avoid having to enter it every time, or program the ATA to allow *97 + mailbox as shortcut that they mention?

Working my way through it, but help from those who might have hit these issues appreciated. Thx.

Re: [Other] VOIP.MS

Thu, 29 Jan 2009 10:00:46 EDT

otty : You shouldn't need a sub-account. My main account SIP URI funstions perfectly. I would ensure all of the settings on your device are correct and that they match the settings on your main account. I had some issues at first setting up SIP on my IP phone. I contacted support and within 12 hours they had researched my device and emailed me the correct settings. Now it works perfectly.

Re: [Other] VOIP.MS

Thu, 29 Jan 2009 09:12:38 EDT

priller : sub-accounts do have an external sip uri. When you create the sub-account, you need to give it an extension number. Then it will create the external sip uri.

It seems like it's better to always use a sub-account, rather than the main account.

Re: [Other] VOIP.MS

Wed, 28 Jan 2009 22:37:06 EDT

Guiness :

said by MartinM :

It will be per main account and sub account.

Main account is always 6 digits. If your account is 100000, the URI would be 100000@sip.us1.voip.ms.

What will be important is using the right server in the URI. If your device is registered with sip.us2.voip.ms, 100000@sip.us2.voip.ms would work, 100000@sip.us1.voip.ms would not.

For subaccounts, it's always Your main account, underscore and an alphanumeric name. In this example, we'll assume the subaccount is 100000_martin.

SIP URI would be 100000_martin@sip.us1.voip.ms

For internal communications, we already allow subaccounts to call each others by the use of internal extension.

For example, you can creat extension 1. Extensions are prefixed with 10, so you would dial 101 to reach that subaccount. It doesn't matter if another customer has 101 has well, we use account code on dialing to not mix extensions.

Are you still around MM?

I can't get the SIP URI for my main account to function - and have been told by support that I need a virtual DID for this. A sub-account, however, does have an external sip uri?

cross-posted to voip.ms forum

Re: [Other] VOIP.MS

Tue, 27 Jan 2009 00:21:54 EDT

otty : Yes I get incoming calls.

I don't see how your computer has anything to do with it. The WIFI connection on your phone works independently of a computer assuming the n95 has a SIP client and is wifi enabled(I am not familiar with that phone).

You need to have an active DID phone number from a voip SIP provider like voip.ms. I could show you the settings you need if you want to set it up.

Edit: A quick google tells me your phone should be capable of voip SIP like mine.

Here are the settings the voip.ms CSR told me to input. First go to menu>tools>settings>connection>SIP>options> create new SIP>default and enter:

Profile: voipms
Service profile: IETF
Default access point: Select your wifi or gprs access point
Public user name: sip:username@sip.ca2.voip.ms
Use compression: No
Registration: Always On
Use security: No

Proxy server:
Proxy server address: sip:sip.ca2.voip.ms
Realm: sip.ca2.voip.ms
User name: username
Password: password
Allow loose routing: Yes
Transport type: UDP
Port: 5060

Registrar server:
Registrar serv.addr.: sip:sip.ca2.voip.ms
Realm: sip.ca2.voip.ms
User name: username
Password: password
Transport type: UDP
Port: 5060

Ensure ther server you enter matches the on you have set on your DID.

Re: [Other] VOIP.MS

Tue, 27 Jan 2009 00:02:24 EDT

anon : Hi otty, did you get incoming calls when you were using your E51?
I tried voip.ms with my n95 nam but however I couldn't get any incoming call.
btw I have my HP with Vista, AMD Turion 64 X2 TL-64 Dual-Core @ 2.2GHz; 4GB RAM since last September :)

Re: Porting Out Question

Sun, 25 Jan 2009 21:25:41 EDT

N9MD :

said by burgerwars :

Now all I need is to learn how to talk fast.
"Hello. Meet you at 6:00. Bye."

If you keep the teeth together, lips slightly apart and fixed, the tongue can be moved very very rapidly .... saving scads of wampum.

Will those with dirty minds please keep your interpretation of my description to yourselves! :uhh:

Re: Porting Out Question

Sun, 25 Jan 2009 21:02:29 EDT

burgerwars :

said by N9MD :

said by burgerwars :

...Any incoming calls I've got (just one) have been wrong numbers. Even though it's just a fraction of a cent to answer these, I still don't want to pay for other people's wrong numbers.

No time out or busy signal or audio problems with my Voip.ms account registering with Dallas on my PAP2T-NA (set at 3600).

As for wrong numbers, if I am correct, MS charges incoming calls in 6-second intervals rather than a full minute. So an 11 second incoming wrong number costs me 0.0027 cents (that's about ¼ of a US penny).

You're correct. If one is on a tight budget and can complete a call in under six seconds (actually less than that since it takes a couple more seconds for their network to see you hung up), one call may only cost about 0.0015. Now all I need is to learn how to talk fast.
"Hello. Meet you at 6:00. Bye."

Re: Porting Out Question

Sun, 25 Jan 2009 20:54:20 EDT

N9MD :

Re: [Other] VOIP.MS

Sun, 25 Jan 2009 20:00:00 EDT

priller :
Just rambling here .....

But, I fine myself asking ... Why does anything related to voip.ms have to go in this one big mega-thread?

A new topic for new issues/questions would make it so much easier to follow.

Re: Porting Out Question

Sun, 25 Jan 2009 19:41:11 EDT

PX Eliezer :

said by burgerwars :

One thing that has happened a couple of times when testing is when I dial my POTS line from VOIP.MS and hang up, my home phone will still ring for several minutes even though the VOIP.MS has disconnected and is ready to make/receive other calls.

I have had this on various Voip services but it always turned out to be the telephone device that was plugged into the POTS line. Uniden phones particularly seemed prone to this.

Also, does the ATA device itself indicate that the call has been terminated? If not, might this be related to a configurable ATA setting? Such as ring debounce, or detect polarity reversal?

Porting Out Question

Sun, 25 Jan 2009 18:41:43 EDT

burgerwars : Not that I plan on doing it any time soon, has anybody had any issues/experience porting out a VOIP.MS issued number to use with another provider? The number I have with them shows in www.telcodata.us as a 360Networks number.

The problem with my VOIP.MS number being busy when it wasn't seems to have fixed itself. One thing that has happened a couple of times when testing is when I dial my POTS line from VOIP.MS and hang up, my home phone will still ring for several minutes even though the VOIP.MS has disconnected and is ready to make/receive other calls. I do like VOIP.MS web interface. Lots of things there for me to futz around with. I'm using their per minute plan and not their incoming unlimited plan. I'm not letting my number failover into voicemail as I haven't given in out to many people. Any incoming calls I've got (just one) have been wrong numbers. Even though it's just a fraction of a cent to answer these, I still don't want to pay for other people's wrong numbers.

Other than that it looks like a good service. I hope they don't vanish.

Re: [Other] VOIP.MS

Fri, 23 Jan 2009 10:06:54 EDT

PX Eliezer :

said by wgraz :

I have been trying to get the Graphical Usage Reports to load for 5 days now.

I had never tried the Graphical Usage Reports before, but I tried them just now.

Works fine.

--------------------------------------

Flash Player 10.0.12.36

Shockwave 11.0.3r471

Mozilla Firefox 3.0.5

Software firewall/HIPS: Online Armor Free 3.0.0.190

Router w/ firewall: D-Link DGL-4100.

--------------------------------------

Hope this helps.

Re: [Other] VOIP.MS

Fri, 23 Jan 2009 09:47:31 EDT

kaila : wgraz- The 'daily spending' and 'calls-and-minutes' reports are occasionally timing out, but the other tabs on the graphics report are working fine for me. This is with FF 3.0.5, flash 10 player, mac OS. If you use FF, are you using adblock or no-script extensions? It's pretty easy to accidentally block the flash panel with adblock, and no-script requires what amounts to site whitelisting before allowing embedded flash to run.

Re: [Other] VOIP.MS

Fri, 23 Jan 2009 08:59:09 EDT

anon : I have been trying to get the Graphical Usage Reports to load for 5 days now. I upgraded my flash to 10(Youtube always played nicely with vers 9), I have tried disabling my popup stopper and software firewall. Priller, what version of Flash player and browser are you using? I cannot figure this out. The negativities are stacking up.

Re: [Other] VOIP.MS

Fri, 23 Jan 2009 00:43:44 EDT

Yippz Voip :

said by otty :

Any ideas as to what was causing the cut-outs in the voice.

Edit: the problem appeared worse when both of us would speak at once.

Sounds like you have the device in simplex mode - see if you can get the sound card or sound device in the phone to use duplex mode so the mic and speaker are open at once. Using my Axim as a sip phone, software like SJPhone would automatically use either the mic or speaker, not both at the same time, to avoid echo. This switching back and forth in Windows CE would cause audio cut-outs. There should be a setting in the phone or in the voip software.

Re: [Other] VOIP.MS

Fri, 23 Jan 2009 00:26:55 EDT

otty : Hi guys.

Just made my first call on my new Nokia E51 using its integrated SIP stack. The clarity was excellent with no echo or delay. There was however intermittent cut-outs in the voice on both ends.

I am registered to the Toronto server using the value routing. The call was made from here in Saskatoon to a small town in central BC.

Any ideas as to what was causing the cut-outs in the voice.

Edit: the problem appeared worse when both of us would speak at once.
--
Sasktel HighSpeed Extreme , Vista, AMD Turion 64 X2 TL-60 Dual-Core @ 2.0GHz; 2GB RAM

Re: [Other] VOIP.MS

Thu, 22 Jan 2009 07:58:52 EDT

hockeynomad : I have sync only as I have a 100' long phone cord from a jack in a new box installed by Bell to extends inside to my modem.

My DSL provider tells me there must be some inside wiring problem. I've heard that a couple of times before as have had same issue, then sync inside mysteriously reappears.

Regarding inside wiring, months ago I paid an independant tech $350 to make wiring compatible for VOIP. So that shouldn't be a problem. Changed the wire from inside demarc jack to modem, no change.

So its that 100' cord for now.

Any ideas on this one?

Re: My VOIP.ms issue.

Wed, 21 Jan 2009 20:39:27 EDT

burgerwars :

said by otty :

I just had an issue that my phone wasn't registered when an incoming call came too (fortunately I have a ring group set up with my cell). How do you set registration intervals? On the voip.ms site or on the SIP device?

Edit: My non-registration could have been due to my connection being maxed out as I was downloading a video.

On the SIP device. They suggest it be set to 3600 seconds on a Sipura, but whatever I set it doesn't seem to matter. On their site you can see if your device is registered, which server and when is the next registration.

I just switched servers to Dallas. It seems the busy signals come less often, but it could be my imagination. I've set no sort of failover yet in their interface.

I'm going to call it a day. I'll look at it this again tomorrow.

This weekend I'm going to try it on a different ATA. I'm thinking it might have something to do with an old Sipura 2000 I'm trying to configure it on line 2/port 5061 with.

Re: My VOIP.ms issue.

Wed, 21 Jan 2009 20:26:22 EDT

otty : I just had an issue that my phone wasn't registered when an incoming call came too (fortunately I have a ring group set up with my cell). How do you set registration intervals? On the voip.ms site or on the SIP device?

Edit: My non-registration could have been due to my connection being maxed out as I was downloading a video.

My VOIP.ms issue.

Wed, 21 Jan 2009 19:55:57 EDT

burgerwars : I decided to give VOIP.ms a try today. I put it on line 2 of a Sipura which was previously used by Stanaphone. I followed their configuration sample and decided to use their L.A. server. It does register and have been able to call. But I would say for about 30 seconds every minute calls can't be dialed and incoming calls will get a busy even though the line isn't. Setting the registration interval real low (like under 30 seconds) doesn't help. Any ideas?

Re: [Other] VOIP.MS

Wed, 21 Jan 2009 15:33:44 EDT

dalrun : My PC > switch > bridged Cisco 678 > ISP gateway is 53ms.

Re: [Other] VOIP.MS

Wed, 21 Jan 2009 14:42:15 EDT

PX Eliezer :

said by hockeynomad :

Can a ticket opened by a VOIP provider cause a Bell Technician to check my wires at my home?

That is not a realistic scenario.

said by hockeynomad :

Since this ticket was opened, I have no DSL and VOIP.

As priller said, you need to take a deep breath and think about the cause and effect relationship.

It is doubtful in the extreme that a Voip provider caused your DSL to die.

It is FAR, FAR more likely that your DSL problems were happening first, and led to the Voip problem.

Now, if Voip.MS has unresponsive customer service, that's a very valid gripe. But they did not trash your DSL.

-----------------------------------------

Occam's Razor has been expressed many ways. Isaac Newton said:
"We are to admit no more causes of natural things than such as are both true and sufficient to explain their appearances."

But a simpler way is:
"When you have two competing theories that make exactly the same predictions, the simpler one is the better."

Re: [Other] VOIP.MS

Wed, 21 Jan 2009 14:10:58 EDT

priller :

said by hockeynomad :

Since this ticket was opened, I have no DSL and VOIP?

I now have my DSL provider, Teksavvy, involved.

Way to go, voip.ms

Take a breath. It's rather obvious that the problem isn't with voip.ms and probably never was. They have not caused your DSL to die. Sure sounding like your DSL problems are what was causing your VoIP problems all along.

Re: [Other] VOIP.MS

Wed, 21 Jan 2009 14:01:48 EDT

hockeynomad : No, english or french was probably his 5th or 6th language.

Since Voip.ms refuses to answer my emails and provide an update, my question here is: Can a ticket opened by a VOIP provider cause a Bell Technician to check my wires at my home?

Since this ticket was opened, I have no DSL and VOIP?

I now have my DSL provider, Teksavvy, involved.

Way to go, voip.ms

Re: [Other] VOIP.MS

Tue, 20 Jan 2009 16:28:53 EDT

PX Eliezer :

said by hockeynomad :

Finally did get a hold of phone tech support, and the guy sounded like he was in a tunnel, or speaking from Pluto...

That's a downer. :(

said by hockeynomad :

and english was definitely not his first language.

In this particular case, wonder if it was French? Because they are based in Quebec...

Re: [Other] VOIP.MS

Tue, 20 Jan 2009 15:55:23 EDT

hockeynomad : Finally did get a hold of phone tech support, and the guy sounded like he was in a tunnel, or speaking from Pluto, and english was definitely not his first language.

Why don't I use Livechat he asked? Well I am still waiting a response.

Keep you posted.

Re: [Other] VOIP.MS

Tue, 20 Jan 2009 10:03:24 EDT

PX Eliezer :

said by hockeynomad :

Voip.ms phone support 877-786-4767 leaves a lot to be desired. Been twice on hold interminably until they drop the line informing the caller to try again later.

You have every right to be ticked off at Voip.MS.

I think that this is a real issue.

And note that yesterday was NOT a holiday in Canada where they are based.

-----------------------------------------------

I am shifting more usage back to CallCentric. Yes, the minute charges are higher, but the support is excellent. Do be aware that their support is by e-mail type system, but, regardless, CallCentric support is quick and responsive.

Re: [Other] VOIP.MS

Tue, 20 Jan 2009 09:49:44 EDT

hockeynomad : Voip.ms phone support 877-786-4767 leaves a lot to be desired. Been twice on hold interminably until they drop the line informing the caller to try again later.

The sole lonely Maytag support guy who told me he only gets 5 calls per shift and hence an immediate response maybe could have been a better VOIP choice. But I went for the lower price....

I had a ticket for my troubles and shortly afterward have had no VOIP since saturday, Jan 18. This 24 hour response just doesn't cut it.

Re: [Other] VOIP.MS

Mon, 19 Jan 2009 11:04:45 EDT

priller :
Working for me .. came up fast.

Re: [Other] VOIP.MS

Mon, 19 Jan 2009 11:02:32 EDT

anon : Is anyone else having issues with the Graphical Usage Reports in the Voip.ms portal not loading. They are timing out showing a large orange rectangle. Thanks

Re: [Other] VOIP.MS

Sat, 17 Jan 2009 14:22:03 EDT

anon : I would think if they are slow to respond, they don't have an answer for ya just yet. I have no idea what causes dropped calls. Router settings maybe?

Re: [Other] VOIP.MS

Sat, 17 Jan 2009 12:00:04 EDT

hockeynomad : I have had a ticket created by voip.ms for some outgoing calls that drop after a few seconds.

They are slow to respond if they do, so what could be causing this?

Gizmo5 DID numbers

Mon, 12 Jan 2009 23:46:26 EDT

PX Eliezer :

said by kieranmullen :

Could you post the full numbers in question? I am curious.

I do not doubt they have different prices for different providers. I just could not any of the higher prices you mentioned.

I looked just before I posted.

Searches at different times seem to yield different results! But I am attaching two screenshots where it can be seen.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 23:09:07 EDT

kieranmullen : Could you post the full numbers in question? I am curious.

I do not doubt they have different prices for different providers. I just could not any of the higher prices you mentioned. I am looking at the full list of Portland numbers, which actually seems quite short, but none of them are $99. How long ago did you look? Also which are codes do you actually want? It seems you are all over the place. I doubt they sold that quickly.

As far as the disparity, I have no idea. Perhaps they choose not to average the cost to avoid getting screwed. Perhaps also the reason why the upped the rate from .01 to .02 per minute for long distance a while back as some carriers charge more to terminate than others.
--
KieranMullen »360oregon.com Betamax/Voipbuster Reseller »infocalls.net/vouchers
IP/Website Tools »ip.drlinky.com

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 22:54:30 EDT

PX Eliezer :

said by kieranmullen :

gizmoproject.com offers dids for $35 a year with *unlimited call in.

I was looking tonight at getting one of those Gizmo5 DID's.

Yes, many of them are offered for $ 35 a year, but many others are priced higher.

Specific lookups on Gizmo5 DID ordering page:

Area Code 848 (NJ): $ 99 /year

Area Code 541 (OR): $ 99 /year

Area Code 503 (OR): DID's variously available for $ 35, $ 64, $ 66, and some for $ 99

I have nothing against Gizmo apparently needing to charge different prices in different areas, but when some of these DID's are $ 35 and some are $ 99 that's a pretty big difference that needs to be highlighted.

Just like years ago when Jay Leno showed an ad from a car repair facility that advertised cheap rates for a tuneup on a 3 (three) cylinder car---Extra charge for more cylinders.

I can't think of any other DID supplier with a 3-1 price differential.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 22:36:15 EDT

priller :
FWIW ... I ported from Vonage to voip.ms and it took exactly 3 weeks.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 22:31:19 EDT

PX Eliezer :

said by kieranmullen :

One thing to consider is are you transferring a number in? Do they charge for it?

Good point to raise.

From their website:

"Price per number ported to voip.ms: $25"

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 22:20:10 EDT

kieranmullen : One thing to consider is are you transferring a number in? Do they charge for it? I could not find in in their FAQ's.
les.net charges $4 a month per number transferred in with *unlimited call in. gizmoproject.com offers dids for $35 a year with *unlimited call in. This is my second year with gizmo did's and no issues. I have even hooked up my relatives with them.

Excluding telemarketing, calling card businesses etc.

said by jay_rm :

I'm considering opening a voip.ms PAYGO account. These are the monthly charges I've spreadsheeted..anything else or any corrections ??

DID/month USD$1.49
incoming/min $0.0149 (to USA numbers)
outgoing/min $0.0105 (to USA numbers)

no additional fees/taxes/ect

Is this in the ballpark ??

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 19:28:56 EDT

N9MD :

said by Yippz Voip :

Is there any way to change an existing DID plan from the per minute rate to the all-you-can-eat rate without losing that particular DID number?

Yes ... just about every VoIP provider permits changing from one plan to another (e.g., PAYG to Unlimited, Unlimited to PAYG).

ViaTalk has a place on their website ... but other providers may require a simple email or Trouble Ticket or call to Customer Service.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 14:52:06 EDT

priller :

said by jay_rm :

said by priller :

CNAM is optional @ $0.0125 per query

Ahhh - so no incoming name CID. Do they still transmit number CID ?

Yes. You will get CID number, but if you want them to do a dip to get the name, it will cost ya.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 14:18:27 EDT

Yippz Voip :
Is there any way to change an existing DID plan from the per minute rate to the all-you-can-eat rate without losing that particular DID number?

I've looked through the user interface and can't see any way to switch plans...

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 14:13:54 EDT

PX Eliezer :

said by jay_rm :

I'm considering opening a voip.ms PAYGO account. Is this in the ballpark ??

Please also see my post, which appears just two posts above yours. :uhh:

Some credit card companies may charge you a fee to use Voip.MS because they consider it Canadian. This has been discussed previously in these forums.

The Voip.MS DID setup fees vary a tiny bit depending on which plan you choose.

And yes, inbound caller ID (number) is included.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 14:03:04 EDT

jay_rm :

said by priller :

CNAM is optional @ $0.0125 per query

Ahhh - so no incoming name CID. Do they still transmit number CID ?
--
3500/512 5.7 GHz Motorola Canopy Wireless; FoxValley.net
'It looks just like a Telefunken U47 !'

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 11:55:58 EDT

priller :

said by jay_rm :

DID/month USD$1.49
incoming/min $0.0149 (to USA numbers)
outgoing/min $0.0105 (to USA numbers)

One time Setup Fee: $0.50 per DID

CNAM is optional @ $0.0125 per query
E911 is optional @ $1.50 per DID per month

If you chose premium outgoing USA route, it's $0.0125

... that's it!

Note that billing is in 6 seconds increments, many providers do only full minute billing. Unlimited channels.

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 11:50:52 EDT

jay_rm : I'm considering opening a voip.ms PAYGO account. These are the monthly charges I've spreadsheeted..anything else or any corrections ??

DID/month USD$1.49
incoming/min $0.0149 (to USA numbers)
outgoing/min $0.0105 (to USA numbers)

no additional fees/taxes/ect

Is this in the ballpark ??
--
3500/512 5.7 GHz Motorola Canopy Wireless; FoxValley.net
'It looks just like a Telefunken U47 !'

Re: [Other] VOIP.MS

Mon, 12 Jan 2009 09:55:38 EDT

kaila : For those with one-way audio and/or drop issues, which routing (vale/premium) are you using, what server are you peering with, and is anyone using IAX? These issues are often notoriously difficult to resolve, and are often local network or ISP issues. Maybe we can uncover a pattern here.

I've had a positive experience with voip.ms since I started using them over the last 6 weeks or so. There has been a very occasional one-way problem (always where the outside party can't hear me). This has also happened in the past with Telasip & Voicepulse as well, so I've considered this issue (very occasional) to be a local network problem.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 21:18:20 EDT

PX Eliezer :

said by wgraz :

What types of fees/taxes does CC hit you with? What about Voip.ms? Thanks

CallCentric:

A one-time setup fee of $1.50 and a monthly fee of $1.50 apply to each Callcentric account requiring 911 service (all US accounts).

One-time setup fee if you purchase a DID.

If customer has an address in NY State, then 8.375% NY State Sales Tax is collected on all billed calls and services. Again, that's only if customer is in NY State.

-----------------------------------------------------

Voip.MS:

Monthly 911 fee of $ 1.50 (optional because they are Canadian).

One-time setup fee if you purchase a DID, about a dollar.

Some US credit card companies impose payment surcharge because they are a Canadian company.

Small extra fee for CNAM, optional.

------------------------------------------------------

Obviously with both companies you have to pay for:

DID number if you get one.
Inbound calling costs.
Outbound calling costs.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 21:00:34 EDT

priller :

said by wgraz :

What types of fees/taxes .... What about Voip.ms?

$0.00

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 20:35:46 EDT

anon : What types of fees/taxes does CC hit you with? What about Voip.ms? Thanks

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 17:33:25 EDT

PX Eliezer :

said by wgraz :

I just wish voip.ms had contact list to enhance the Caller ID experince like CC.

It's tough. Different services offer different features, rates, and reliability. Kind of like choosing which mate is best.

I use Voip.MS as the "private" line at my office. Recently has seemed fine for incoming but we have had dropped calls and one-way audio on outgoing. This may have improved somewhat after I switched from NY server to Dallas server.

I think that Voip.MS is decent overall, has great potential, and some interesting features. But I do think that Voip.MS cannot match CallCentric for customer service. Of course, CC has somewhat higher rates, so that's a tradeoff.

And clearly Voip.MS needs to get a handle on these recent issues that many people have raised.

I use CallCentric as primary home line, as I just have more faith in their excellent long-term track record, and personal excellent experience with their customer service.

I would try VOIPo---and still might---but I like to be in control of my own VoIP hardware.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 17:30:05 EDT

gg22 : If you have two devices with voip.ms you should open two sub-accounts and register each device with different sub-account. Then use multiring feature to ring both sub-accounts. That's the way it's set up to work.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 17:04:05 EDT

anon : OK...long story short.... I had two devices (pap2 and pap2t) both connected and configured with my voip.ms name and pw. That was mostlikely causing dialing issues. I don't know about the dropped calls. Time will tell.
I just wish voip.ms had contact list to enhance the Caller ID experince like CC. Sorry for so much posting :)

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 16:23:34 EDT

priller :

said by wgraz :

So by now, I'm tired of contacting them and, they don't have support on the weekend anyways.

They have weekend support via email, just no phone support.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 16:16:15 EDT

anon : yes, that's how it is for me also. the dropped calls sounds like each person just stopped talking. Dead air...no idal tone. I hung up the phone after a minute or so after being dropped.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 16:11:26 EDT

anon : I have contacted support on a number of other issues:
1. Post dial delay - They sent me a dial plan that resolved the problem.
2. I could not speed dial to VM. PX Eliezer helped me with this and support has not gotton back to me on this.

And since this past Fri, two more issues have come up:
When the caller dials me, the phone is ringing on her end and then goes to VM. My phone never rings.

And the dropped calls. I've only been a customer for a week or so. I'm using the NYC server(I live about 50 miles away. So by now, I'm tired of contacting them and, they don't have support on the weekend anyways. I was just recently thinking of thowing in the towel. I just might stick with Voicepulse for $17/month includ taxes/fees. I had VP for 3 years but I was looking for a cheaper plan.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 15:55:44 EDT

priller :

said by wgraz :

I already had some dropped calls also. I don't get a dial tone when the call is dropped - just dead air. I placed the pap2 address on DMZ and we'll see how that does.

Please, please let voip.ms support know about this!!!

I've been using them since July with zero problems. However, in the past couple of days, numerous calls have been dropped. Neither party can hear each other after a random period of time (always less than 15 minutes). I've had the same problem registered to Dallas and NY. Nothing has changed in my setup and other VoIP providers are not affected.

Is that the symptom you have had?

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 15:28:13 EDT

anon : Thanks. I hear ya. No Voip co has all. Why did you choose CC and why didn't you choose voip.ms for the home? I must say, I'm getting more and more disenchanted with Voip.ms.
I already had some dropped calls also. I don't get a dial tone when the call is dropped - just dead air. I placed the pap2 address on DMZ and we'll see how that does.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 15:20:34 EDT

usa2k : VOIPo used to greet with Comedian Mail - part of asterisk.
I don't think VOIPo uses asterisk for VM any more.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 14:51:43 EDT

PX Eliezer :

said by wgraz :

Are the NAT settings(NAT Mapping Enable and NAT Keep Alive Enable) supposed to be set to yes if your behind a router using a Pap2?

Yes on both.

said by wgraz :

I wish that Voip.ms had contact lists so one can place names and numbers of cellphone callers so the name shows in the Caller ID.

This feature is available on CallCentric---but on the other hand, CallCentric does not currently offer regular CNAM. I kind of like CallCentric's method better, at least for residential use.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 14:31:10 EDT

anon : Are the NAT settings(NAT Mapping Enable and NAT Keep Alive Enable) supposed to be set to yes if your behind a router using a Pap2? Sporatically, when others call me, it goes to VM(after 40s) and my phone never rings. I read that it has something to do with NAT.

I wish that Voip.ms had contact lists so one can place names and numbers of cellphone callers so the name shows in the Caller ID. Has anyone found a work around for this? My phone has a phonebook but it does not bring up a phone book entry when that person calls.

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 14:03:26 EDT

PX Eliezer :

said by wgraz :

Actually I just took out the 9 like: *xx. and still works! Thanks so much!

You're welcome.

Shows the problems that can occur when someone misses a period. :)

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 14:00:46 EDT

PX Eliezer : Neat story! :)

However, I believe that "Comedian Mail" is associated with the Asterisk system.

Why Asterisk chose the name Comedian Mail, I have no idea.

The only thing I can think of is, were they thinking of Buddy Hackett?

.

Hackett meaning HackIt

Re: [Other] VOIP.MS

Sun, 11 Jan 2009 13:11:33 EDT

amhutap : "Why they call it Comedian Mail, I don't know....."

I'll take a stab at it, Voip.MS is (part?) Canadian based, and once upon a time pride of Voicemail systems here was (or was it all over North America?) Nortel Meredian systems.

The way it went was: You got a 'proper' job, you got a voicemail password from HR on an official looking form, you hit VM on your Nortel desk phone and first thing you heard was "Meridian Mail" in a calm, confident, almost authoritative female voice. Suddenly ground would go solid underneath you feet and you knew you were working for the man. That of course all is before Nortel got where they are today and were the premium traditional PBX with costs to match.

My guess is that some or many of Voip.Ms people grew up listening to "Meridian Mail" setting up their VM boxes and leaving VMs for other who had 'real' jobs and now can't help but smirk...

My $.02 worth...

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 18:59:13 EDT

anon : Great! Thanks! Now it works like it's supposed to- using *97XXXXX

Now when dialing *97##XXXX I get the "Comedian Mail" prompt.

Actually I just took out the 9 like: *xx. and still works! Thanks so much!

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 18:39:37 EDT

PX Eliezer :

said by wgraz :

Using a pap2. I tried *9x in the dial plan...no dice.

Again, I don't know if it will help or not, but just in case, I want to make sure you get every opportunity.

It's NOT *9x but rather *9x. with a PERIOD at the end.

Good luck.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 18:22:33 EDT

anon : Using a pap2. I tried *9x in the dial plan...no dice. I just don't understand why dialing a 5 digit mailbox number to access VM is mandatory. Why make things difficult.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 16:47:46 EDT

PX Eliezer :

said by wgraz :

Here's the correct dial plan....

What type of ATA or other device are you using? Syntax differs among manufacturers.

One might think that the *xx in the above dial plan should suffice, but you never know.

So I would again suggest: Try inserting *9x. if using a Linksys/Sipura device, or it might be *9x+ on some other systems. Note that it needs to be within a pair of | but otherwise it doesn't matter where you put it in. Note that the x is followed by a period or by a + depending on manufacturer.

If this solves your voicemail issue, great. If not, well, you've wasted 5 minutes, and I'm sorry.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 16:17:59 EDT

anon : Posting to this forum is trunkating the beginning of my dial plan. sorry

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 16:15:15 EDT

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 16:13:24 EDT

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 16:05:05 EDT

anon : But if one uses a call forwarding number for Failover, by default, call forwading to that number is will always be enabled, correct?

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 15:50:29 EDT

PX Eliezer :

said by wgraz :

Yes, both are set to NYC. No go. Like I said, when I enter *97, Mailbox, Password, one at a time, I can access VM but before I do that, I will cancel service.

Does your dial plan include the syntax that is needed for this sequence?

That is, *9x. if using a Linksys/Sipura device, or *9x+ on some other systems?

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 14:50:41 EDT

anon : Thanks! I'm starting to understand beauty behind this seemingly raw html interface.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 14:28:24 EDT

priller : Configure the external number under "DID Numbers --> Call Forwarding" (you're just creating an object). Then it will be available in the Failover pulldown under "Edit DID Settings".

The same logic applies to Ring Groups and Time Conditions.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 14:05:53 EDT

anon : How do you set to external number? I was in "Manage Dids" and the Failover setting only gives a pulldown menu. Thanks

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 13:51:23 EDT

priller :

said by wgraz :

What about Failover...can't I enter an external number like a cell phone number- Strange.

I have failover set to an external PSTN number. Tested and works fine. Make sure you enter it with a leading 1.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 13:17:31 EDT

anon : Yes, both are set to NYC. No go. Like I said, when I enter *97, Mailbox, Password, one at a time, I can access VM but before I do that, I will cancel service. I put a ticket into support on Thurs but haven't heard from them yet...they normally respond within 24 hours. Do you guys get the "Comedian Mail" prompt? What about Failover...can't I enter an external number like a cell phone number- Strange.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 13:01:28 EDT

PX Eliezer :

said by Yippz Voip :

No problem... I have my voicemail access entered into my speed dial simply (through trial and error) using this method:
*97PPXXXXXX
P=Pause (be sure to enter 2 of them, usually using the # key when programming your phone)
X=Your voip.ms mailbox

The only prompt you'll hear now is "access code"
You will still have to enter your secret mailbox access code, which is still a good idea.
Quick, easy and works everytime...

(See edit at bottom).

As an extension to the above, you can also do this which includes everything:

*97ppXXXXXXpppAAAA

where p = pause, X = Mailbox, and A = Access Code/Password.

But of course do NOT set this up with your access code unless your phone is totally secure.

And yes, it IS "Comedian Mail", not to be confused with "Chameleon Mail" which is a different service.

Why they call it Comedian Mail, I don't know.....

------------------------------------------------

George Carlin: "What if there were no hypothetical questions?"

------------------------------------------------

EDIT: I posted this before the post from wgraz indicating that it still was not working. I think that priller's solution may be the issue.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 12:48:36 EDT

priller : Make sure you are registered to the same server you originally created the mailbox for.

"Please note that voicemail system is not centralized. It is independant and per server. For example, if your DID is currently on POP US3 (Los Angeles) and you dial *97 on US4 (New York), you will not access the same mailbox. The same issue can occur if you change the POP (server) of your DID to another server. If you would like our team to migrate your voicemail messages and unavailable message to another server, please send us an email to support@voip.ms with the details."

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 12:44:55 EDT

anon : Thanks. I already tried that. When I use *97##xxxxx as speed dial, It asks for "password". I enter my password and I get "login incorrect". Can't figure it out. For sure I'm using the correct mailbox and password numbers.
Thanks.

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 12:26:50 EDT

Yippz Voip :
No problem... I have my voicemail access entered into my speed dial simply (through trial and error) using this method:

*97PPXXXXXX

P=Pause (be sure to enter 2 of them, usually using the # key when programming your phone)

X=Your voip.ms mailbox

The only prompt you'll hear now is "access code"

You will still have to enter your secret mailbox access code, which is still a good idea.

Quick, easy and works everytime...

Re: [Other] VOIP.MS

Sat, 10 Jan 2009 12:15:35 EDT

anon : When I access my voip.ms voicemail with *97, the prompt says "Comedian Mial" (what?) and then asks for a mailbox. The only way I can access my VM is entering one at a time after each prompt: *97 then enter my mailbox number and then enter my password. Is there ANY WAY one can speed dial this in? Is there a way not to even use a Mailbox? When I try to speed dial *97w/Mailbox like voip.ms indicates, it prompts for the Mailbox all over again. An ideal situation is the speed dial everything to get the VM direct. With my previous carrier, I had the speed dial setup for *123+pin and it worked great. So basically I'm asking, what is the fastest method possible to access VM with the least amount of dialing?(besides email attachements) Thanks.

Re: [Other] VOIP.MS

Fri, 09 Jan 2009 07:43:25 EDT

hockeynomad : wow I was getting in the 70ms range and up last night on Toronto server and 61 TTL -> whatever that is.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 22:33:34 EDT

amhutap : I am with Teksavvy as well in Toronto region and have a 10ms ping to sip.ca2.voip.ms (Toronto Server, actually in hamilton on MountainCable)

Ping to La Belle Server (sip.ca1.voip.ms) is in the 15 ms range.

As much as toronto server pretty much always works, MTL just works for me, always.

CA2 routes through Torix-->Mountaincable, 5 hops
CA1 routes through Torix-->iweb's link to MTL, 7 hops

Hope this helps.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 22:33:28 EDT

gsar : The G.992.1 near the top means you're already on ADSL1. Your modem reports a latency of 21ms, which seems to be on the higher side. Could be due to quality of your DSL line.

Please read the link I sent to understand how to expose the extra tab in DMT with the options to tweak things. Needless to say, if you tweak things and end up breaking stuff, you are entirely on your own. ;)

If you can convince your ISP to put you on fastpath rather than interleaved mode (see the link for the difference between the two modes) that might help with the latency, but your line quality may not allow it.

Having said all that, I'd advise not changing anything. 21ms is not the end of the world, latency-wise, for VoIP. (If you're a hard-core gamer, you might think different.) If your overall latency to the SIP server is less than 150ms you're doing just fine.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 22:14:41 EDT

hockeynomad : or more details

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 22:05:50 EDT

hockeynomad : DMT stats. How to force ADSL1?

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 20:51:14 EDT

gsar : If the modem is unlocked, you can use the DMT tool to switch to ADSL1. If it is locked, you'd need to either buy your own modem or unlock the provider's modem first by reflashing it (assuming your provider would be ok with this). I'm running the 6.1.0.5 firmware--later versions are less tweakable in some ways but should allow you to switch to ADSL1.

Look in the Canadian forums corresponding to your ISP for more info.

For traceroute, you probably already have it--it is spelled tracert at the Windows command line. Just run it with an external host name as the argument and look at what the latency between your router and your ISP's gateway is. If it is over something like 12ms, you can probably tweak the modem to bring it down, depending on line conditions.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 18:58:40 EDT

hockeynomad : Yup, I have the same 516 modem. How the hell do you force it to ADSL1 and is there a traceroute tool anywhere out there?

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 17:42:24 EDT

gsar : In my experience, much of that 46ms latency could be coming from the DSL modem. I had a Thomson Speedtouch 516 modem (one of the best around, btw) running on an ADSL2+ (interleaved) profile, and the modem reported a 36ms latency to the ISPs gateway. When I forced it to ADSL1, the latency dropped to 4ms. Apparently a known issue with that modem and my service provider.

I'd suggest running a traceroute to see which segment is responsible for the bulk of the latency.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 17:36:54 EDT

Yippz Voip : Looks like there are lots of problems recently with Teksavvy's DSL product...

> »ERX06 latency

Could explain the latency problem...

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 16:53:04 EDT

TJ_in_IL :

said by PX Eliezer :

Yah, that sounds like US Airways which proposed this route to get me from Newark, NJ to Detroit, Michigan:

Newark, NJ
to Charlotte, North Carolina
to Philadelphia, PA
to Detroit, Michigan

But think of all of those miles you earn!

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 16:39:50 EDT

PX Eliezer : Well, everyone's comments have been very interesting, and this has been quite educational. Thanks to all.

said by nitzan :

Basically, Comcast is routing from Chamblee, GA to Smyrna, GA (15 mile apart) via Washington...

Yah, that sounds like US Airways which proposed this route to get me from Newark, NJ to Detroit, Michigan:

Newark, NJ
to Charlotte, North Carolina
to Philadelphia, PA
to Detroit, Michigan

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 16:08:56 EDT

espaeth :

said by nitzan :

What you CAN judge by is through where you're being routed. i.e. Atlanta->Dallas directly is better than Atlanta->Washington->Dallas even if the latter has less hops.

That routing is a matter of common points of interconnection between the sites. The trace you showed above shows a route through Washington D.C. because XO doesn't have the necessary network infrastructure to effectively peer with Level(3) in Atlanta. XO's network map shows the only nearby peering points are Dallas, Wash DC, and Miami. From a routing standpoint Dallas and Wash DC will get priority for routes simply because there is significantly more peering density in those locations than Miami.

You can live with a freeway running through your back yard and still need to drive a mile to get on it, depending on how far out entrance ramps are; it's the same scenario here.

said by nitzan :

"multiple locations means better quality" is a myth. It usually doesn't matter inside the US.

On this point we agree. Between any connection points in the continental 48 your audio delay will still be lower than the average cell phone call. There are 2 network metrics that determine 99% of voice quality: jitter and packet loss. Latency simply doesn't matter for most purposes.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 15:58:59 EDT

hockeynomad : Hops? isn't that something they use to make beer?

Seriously, here is my response from Voip.ms Martin:
"Hi! There's virtually no difference between 46 ms and 15 ms. This is
0.015 second vs 0.040 second. You may have more than latency issues if
that's what you are experiencing with our service. This type of issue
can be caused by variable latency (going up and down), bandwidth
instability, bandwidth limitation and jitter.

If you can provide a small sample of calls that had these issues, we will investigate. You can also try another server, like Montreal to see if the route is better between your internet connection and the server."

My pings to Montreal server were around 55ms and the New York 72ms.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 15:49:00 EDT

nitzan : More hops is not always a bad thing. A lot of "hops" are simply multiple routers and switches at the same data center. You could have an IP carrier that switches you over a few hops inside their data center, but your traffic is going over 10Gig fiber, while another IP carrier will bounce you through less hops in the data center - but send you over a 1Gig fiber. Essentially when it comes down to it you cannot judge by simply number of hops.

What you CAN judge by is through where you're being routed. i.e. Atlanta->Dallas directly is better than Atlanta->Washington->Dallas even if the latter has less hops.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 15:42:59 EDT

nitzan :

Not unusual if you use Comcrap.....

Tracing route to atlantapd.org [67.106.220.137]
over a maximum of 30 hops:

1 2 ms 1 ms 1 ms 192.168.1.1
2 * * * Request timed out.
3 15 ms 22 ms 9 ms ge-2-6-ur01.d9chamblee.ga.atlanta.comcast.net [68.85.173.181]
4 9 ms 9 ms 9 ms te-9-1-ur02.d9chamblee.ga.atlanta.comcast.net [68.85.232.46]
5 8 ms 9 ms 11 ms te-9-1-ur01.d8decatur.ga.atlanta.comcast.net [68.86.106.146]
6 9 ms 17 ms 10 ms te-9-1-ur02.d1stonemtn.ga.atlanta.comcast.net [68.86.106.150]
7 15 ms 9 ms 9 ms te-9-1-ur01.d1stonemtn.ga.atlanta.comcast.net [68.86.106.154]
8 9 ms 9 ms 10 ms po-4-ar01.d1stonemtn.ga.atlanta.comcast.net [68.86.106.158]
9 16 ms 21 ms 19 ms po-2-ar01.b0atlanta.ga.atlanta.comcast.net [68.86.106.6]
10 14 ms 12 ms 11 ms te-0-1-0-4-cr01.atlanta.ga.ibone.comcast.net [68.86.90.121] 11 43 ms 201 ms 203 ms te-3-3.car1.Atlanta2.Level3.net [4.71.252.1] 12 13 ms 13 ms 22 ms ae-62-51.ebr2.Atlanta2.Level3.net [4.68.103.29] 13 11 ms 18 ms 16 ms ae-63-60.ebr3.Atlanta2.Level3.net [4.69.138.4] 14 39 ms 35 ms 37 ms ae-2.ebr1.Washington1.Level3.net [4.69.132.86] 15 27 ms 35 ms 37 ms ae-61-61.csw1.Washington1.Level3.net [4.69.134.130]
16 29 ms 25 ms 25 ms ae-21-69.car1.Washington3.Level3.net [4.68.17.7]
17 27 ms 28 ms 25 ms XO-COMMUNIC.car1.Washington3.Level3.net [4.71.20
4.26]
18 46 ms 27 ms 28 ms 65.106.1.114.ptr.us.xo.net [65.106.1.114]
19 28 ms 28 ms 28 ms 65.106.1.113.ptr.us.xo.net [65.106.1.113]
20 44 ms 40 ms 41 ms p6-0-0.rar2.atlanta-ga.us.xo.net [65.106.0.6]
21 41 ms 41 ms 41 ms p4-0-0.MAR2.Smyrna-GA.us.xo.net [65.106.4.18]
22 44 ms 51 ms 43 ms ge13-0.clr1.smyrna-ga.us.xo.net [207.88.85.22]
23 47 ms 45 ms 45 ms 65.104.202.110.ptr.us.xo.net [65.104.202.110]
24 46 ms 52 ms 47 ms 67.106.220.130.ptr.us.xo.net [67.106.220.130]
25 52 ms 63 ms 60 ms 67.106.220.137.ptr.us.xo.net [67.106.220.137]

Trace complete.

Basically, Comcast is routing from Chamblee, GA to Smyrna, GA (15 mile apart) via Washington....

This is not uncommon, in fact it's VERY common for some ISPs. Just because you're physically close to a server doesn't mean you're network-close to that server. In fact from a network perspective I am apparently closer to Future Nine's servers in Miami than to Smyrna which is a 10 minute drive away - because my packets to Future Nine go directly Atlanta->Florida rather than Atlanta->Washington->Atlanta.

For that reason I think having multiple geographic locations is typically meaningless. A server in Miami will do just as well as a server in New York and vice versa. The ONLY meaningful difference is between an east coast and west coast servers. i.e. a server in Miami would perform better than a server in California if you live in Atlanta.

Having said that- even if you live in Seattle, a server in Miami will still have decent quality if everything else is in place. "multiple locations means better quality" is a myth. It usually doesn't matter inside the US.
--
Nitzan Kon, CEO
Future Nine Corporation

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 15:40:11 EDT

Yippz Voip :

said by PX Eliezer :

Nitzan, not trying to be querulous here, and I defer to your expertise. But don't you think it's unusual to have a 46 ms ping between places that are about as close as (in Georgia terms) Marietta and Atlanta?

I know this is an old discussion. On the one hand, some folks say that ping is not the big thing. On the other hand, various providers DO seem to set up servers in multiple geographic locations, presumably for quality and not just redundancy.

...and what about hops?? I'm in Atlanta and connect to the Dallas server with ping times around 29-33ms on average and with about 14 hops -- however, when I ping the New York server I get ping times around 21-29ms on average with 21 hops. BTW, both cities are almost equal distance from ATL according to Google maps. So even with the slightly higher ping times to Dallas, I assume less hops means better quality.

So ----- how does this affect voip and could this help the OP in Mississauga?? Maybe if he tried the MTL server his ping times would increase but he could have fewer hops??

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 15:00:14 EDT

PX Eliezer : Nitzan, not trying to be querulous here, and I defer to your expertise. But don't you think it's unusual to have a 46 ms ping between places that are about as close as (in Georgia terms) Marietta and Atlanta?

I know this is an old discussion. On the one hand, some folks say that ping is not the big thing. On the other hand, various providers DO seem to set up servers in multiple geographic locations, presumably for quality and not just redundancy.

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 13:12:19 EDT

nitzan : 46 ms is by no means high. Sure, it's not uber low - but it's not abnormal and you should have excellent call quality with that unless you are losing packets and/or have jitter which I doubt you have.

Echo is usually created at the TDM end - or in other words whoever voip.ms is using to terminate calls is causing the problem. Probably just bad configuration on their TDM gateway. voip.ms should contact their terminating partner and ask them to fix it.

You could also reduce echo a little by playing with your settings. In the regional screen (assuming PAP2) change:
More Echo Suppression: yes

and if that doesn't help try to reduce the gain values (same screen). This will make the volume lower which results in less echo (but you can't hear 'em, either).
--
Nitzan Kon, CEO
Future Nine Corporation

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 11:40:53 EDT

hockeynomad : my DSL downloads are consistently around 2.5 - 2.8 MB though its been awhile since I ran a speedtest.

Could this profile cause the high ms?

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 11:10:50 EDT

PX Eliezer : Hmmmm.

I am in central New Jersey. Think Springsteen and The Sopranos. (New Jersey---the Family state).

My ping time to Voip.MS server in nearby New York City is consistently around 12 ms.

My ping time going to their server in Toronto is consistently about 25 ms!!

So from here in New Jersey, I have better access to the Toronto server than you have from a Toronto suburb!!!

(And my ping to CallCentric's server in New York is 14 ms).

Bugs Bunny said that something was screwy in Saint Louie. To me, it seems that with your connection, there is not enough pronto in Toronto.

Unless this difference is due to DSL being slower?

(Edit: I could not think of anything to rhyme with Mississauga).

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 10:49:39 EDT

hockeynomad : Interesting PX Eliezer,

I chatted with voip.ms support and she informed me 46ms was "excellent" .

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 10:29:59 EDT

PX Eliezer :

said by hockeynomad :

So I ran some ping tests and the Toronto server is the best at 46ms. Is this satisfactory?

Something seems VERY wrong here!!

Your profile says that you are in Mississauga, Ontario, which is part of the metro Toronto region.

If you are pinging the Toronto server, the ping time should be around 10-15 ms, NOT 46 ms!!!

Can other folks please comment on this too?

Re: [Other] VOIP.MS

Thu, 08 Jan 2009 09:16:51 EDT

hockeynomad : I have some issues with voip.ms

Apparently the other side of the call complain of hearing echos and my voice is choppy and like talking in a tunnel.

At my end though, the voice quality is pristine much better than Ma Bell landline.

Other minor issues are that sometimes on termination calls it takes up to 45 seconds to engage a diatone and sometimes the touchtone just doesn't work.

So I ran some ping tests and the Toronto server is the best at 46ms. Is this satisfactory?

I also ran some other tests and the bounce going out to coming back are nearly identical. Not sure why the other line still encounters the above mentioned problems.

So how to tweak to get these people off my case regarding "my stupid internet phone".

I use Teksavvy's Dry-DSL service. They are a very reputable bunch, but perhaps a problem with the DSL is causing this.

Overall, I am still satisfied with voip.ms especially after the horrendous experience with Unitz.

But things could be better.

Re: [Other] VOIP.MS

Tue, 23 Dec 2008 00:25:18 EDT

N9MD :

said by hockeynomad :

voip.ms customer service can be somewhat disappointing. I can live with the live chat, it is a cost-effective way to handle queries. Even the sometimes 24 hour turnaround is tolerable, but they could do better.

Here's a portion of one of my recent posts specifically addressing the issue of Telephone Support vs. Trouble Ticket Submission.

"As to the lack of telephone support with [fill in the blank] ... some of us prefer on-line Trouble Ticket contacts ... because rather than getting useless scripted "suggestions" from an uninformed Level 1 CSR, a written request can be dispatched to higher tiered individuals who actually have the answers at their fingertips .. so a single response within a few hours (or the following morning for late night trouble ticket submissions) will more often than not resolve any problem quickly."

My current favorites are CallCentric, VOIPo and PhonePower ... and I have never used telephone support. Everyone of my TT submissions with various questions and issues, with each of these vendors, received timely responses and resolutions without the need to sit through a Level 1 CSR's script reading ... followed by hemming and hawing ... leading to confusion and misinformation.

Re: [Other] VOIP.MS

Mon, 22 Dec 2008 16:36:35 EDT

Test99 :

said by hockeynomad :

Is this acceptable?

No.

You might try contacting MartinM. It looks as if he hasn't logged in in the last month, so an IM may not work. But try sending email through BBR.

Re: [Other] VOIP.MS

Mon, 22 Dec 2008 16:31:44 EDT

priller :

said by hockeynomad :

I would prefer a voice contact.

They do have voice support ... 9-5 M-F .... 1.877.7.VOIP.MS

Did you ever try that?

I contacted support via email 3 times. Twice I got a reply in 2-3 hours. The other time it was about 12 hours. Each time the reply was informed and appropriate for the issue/question.

Re: [Other] VOIP.MS

Mon, 22 Dec 2008 16:27:54 EDT

hockeynomad : voip.ms customer service can be somewhat disappointing. I can live with the live chat, it is a cost-effective way to handle queries. Even the sometimes 24 hour turnaround is tolerable, but they could do better.

I would prefer a voice contact. The biggest beef I have is the live chat rep when they can't give you an answer, they sometimes just disappear, hoping you will as well.

Typically, I get the "one moment please", but today I got nothing and waited 10 minutes for a response. I didn't get one. So I just logged out.

Is this acceptable?

Re: [Other] VOIP.MS

Sat, 20 Dec 2008 10:58:18 EDT

PX Eliezer : I see that CallWithUs has at least one Windsor number available at present---

519-800-01xx

(searchable on their website).

CWU does have various specific numbers available in Canada.

-----------------------------------------------------------

Voip.MS of course has many numbers in Canada but as "themew" noted, they don't seem to be searchable for specifics, unlike the Voip.MS USA numbers.

Re: [Other] VOIP.MS

Sat, 20 Dec 2008 06:47:30 EDT

usa2k : I did not pick a specific Windsor Ontario Canada number ... I chose the location, and eventually it was available to me.

Been a while since I did that. I needed some manual approval, so it was a couple days.

Re: [Other] VOIP.MS

Fri, 19 Dec 2008 23:44:47 EDT

Yippz Voip : I see I can choose a number for US and tollfree but I don't see that for International and Canadian numbers.

After you tell the interface you want to order 1 Canadian or Intl number, do a list of numbers come up to choose from or are you simply assigned a number from the pool?

Re: [Other] VOIP.MS

Wed, 17 Dec 2008 23:14:22 EDT

Darkev : Are you or is anyone here familiar with how to add the MWI functionality to the Linksys SPA9000 so that it will work with voip.ms? I've played with this for hours and hours and tried all sorts of different combinations and cannot get it to work.

You help would be greatly appreciated.

Thank you.

Re: [Other] VOIP.MS

Fri, 24 Oct 2008 19:59:31 EDT

usa2k : I see you found it »voip.ms

Re: [Other] VOIP.MS

Fri, 24 Oct 2008 16:59:28 EDT

Snake Doctor : I think it's voip.ms had their own section here on BBR like viatalk and vonage so we could discuss problems and solution there without going through one long thread!!

Re: [Other] VOIP.MS

Thu, 23 Oct 2008 04:21:52 EDT

Snake Doctor : All sorted out now,i'm good to go!

will report back

Re: [Other] VOIP.MS

Wed, 22 Oct 2008 14:55:35 EDT

hockeynomad : were you using Unitz with Teksavvy?

They were horrendous.

I've had a couple issues with voip.ms but i believe that had to do with problem with Toronto servor. I went back to Montreal servor.

The only other issue occured when i was downloading a torrent though it was only 10KB/s at the time.

Yes the voice quality is pristine.

Re: [Other] VOIP.MS

Tue, 21 Oct 2008 11:52:59 EDT

snickersee : Just signed up with voip.ms. They manually verified my account and that took about 30 mins.

Outstanding voice quality and great rates. Just made an overseas call that would have cost 5X as much via my POTS line.

Hopefully they can keep up the great work and hopefully prices will stay the same/decrease over time. I'm a very satisfied customer right now.

Re: [Other] VOIP.MS

Sun, 19 Oct 2008 14:56:24 EDT

gg22 : There was an email from them a few days ago about spike of fraudulent activity. That's maybe why they tightened their security again. According to their rep Martin they do it from time to time if they detect something fraudulent. My set up process took seconds 6 months ago. I'm paying by PayPal.

Re: [Other] VOIP.MS

Sun, 19 Oct 2008 11:09:21 EDT

JohnnyBeGood :

said by Snake Doctor :

I signed up with them yesterday and i got to chat with a rep who was suppose to manually activated my account but said they had some security issue they had to check before i'd be fully activated
He said it was going to take an hour and that i should be getting an email from them with the results of what happens,that is they are going to give me a line or not
I found that really odd and wanted to ask you guys here if you went through the same experiance as me
I'd probably sign up with a different provider today if i don't get any emails from them

Read my post here »800 # with free incoming minutes?
I think you might be going thru the same I went.
After MartinM offered me credit on my account I tried and liked it, not too long ago I ported my # to them. Overall they're good provider. From my experience all what I can say "don't judge a book by its cover".

Re: [Other] VOIP.MS

Sun, 19 Oct 2008 10:34:46 EDT

Snake Doctor : I signed up with them yesterday and i got to chat with a rep who was suppose to manually activated my account but said they had some security issue they had to check before i'd be fully activated
He said it was going to take an hour and that i should be getting an email from them with the results of what happens,that is they are going to give me a line or not
I found that really odd and wanted to ask you guys here if you went through the same experiance as me
I'd probably sign up with a different provider today if i don't get any emails from them

What is Important to me

Fri, 17 Oct 2008 22:18:43 EDT

anon : Importance to me in order of priority

. Decent Tech Support
. Good sound quality and reliability
. Fair Prices
. Phone Conversation Encryption
. Voice to Email with attached message
. Email to fax and fax to email
. Extra 'VoIP' features such as the virtual SIP and all the other stuff you have implemented

Out of this list, the only things that are not satisfied to my level of comfort is Encryption & email to fax and fax to email.

About the encryption. Not sure if the technology really exists to fully satisfy this yet. Can you set up your server to take secure connections and pass them off to my linksys box? :hmm:

Dr. Sassafras

Re: [Other] VOIP.MS

Thu, 16 Oct 2008 12:39:42 EDT

hockeynomad : Hi Martin,

Phase 3 with callerIDBlocking and AnonymousCallerID blocking, mean the most to me.

Any idea on its implementation?

Re: [Other] VOIP.MS

Wed, 08 Oct 2008 20:48:04 EDT

johnny90025 : Does anyone use Voip.ms with a Polycom phone, especially using the phone's local settings as opposed to a web-based configuration file? I have the Polycom 650 and am trying to figure out how to configure it for Voip.ms. If your phone is the same or similar and you could post your settings here or e-mail me screenshots, I will be very grateful! :D :D :D

UPDATE: Voip.ms has excellent tech support via chat, and they got me going after two chat sessions by changing one thing I had set wrong.

Re: VOIP.MS features suggestions

Tue, 07 Oct 2008 13:00:59 EDT

MartinM : I doubt it's the way this ticket has been closed but I'll follow up for you. However it's true there was no issues with the server itself. ~~If you could send me your account, I'll track the ticket. thanks~~

Edit: I was able to track down your ticket. This ticket been assigned to a sysadmin by the person you chatted with for review.

Re: VOIP.MS features suggestions

Tue, 07 Oct 2008 12:26:51 EDT

hockeynomad : Martin, I did have outgoing calls issues and incoming call problems this past weekend.

I used your chat support and informed them the actual 10 digit numbers involved. I am using the Toronto CA2 server.

The only response I got was there were no issues on this server.

Re: VOIP.MS features suggestions

Mon, 06 Oct 2008 21:37:38 EDT

MartinM : You can personalize your greeting, just log in your mailbox and press 0 for options. From there, record your "unavailable" message and it will override "the person at extension X is unavailable".

Virtual Fax by web interface is something we're working on. It will include incoming and outgoing.
It's not ready it's in its early stages so my guess is around 2 months for that to be implemented.

About the other suggestions, thank you for the feedback. It's always submitted for reviews to voip.ms team.

VOIP.MS features suggestions

Mon, 06 Oct 2008 20:23:46 EDT

gg22 :

About incoming call issues, we didn't have any issues that I am aware of so I'll suggest contacting support about it.

There was no problem with voip.ms. I played with new VM features and pressed *78 (block all calls :o) instead of *98 (check voicemail) on my ATA. Took me a whole day to understand what was wrong.

Some of the additional features it would be nice to see in voip.ms:
~~1. Personalised VM greetings~~ (Already Supported)
2. Reduced cost in-network calls (the same way it's done for the calls between extensions.
3. Fax to email service
4. Per call value/premium route selection for N/A calls. (the same way it's done for international calls).

Re: VOIP.MS Bank Credit Card Transaction Fees

Mon, 06 Oct 2008 19:47:44 EDT

wierdo :

said by PX Eliezer :

CapitalOne charges NO fee at present:

That was true up until about a year ago. I used my Capital One card for the first time in years last time I was in Canada, the summer before last. A few months after that trip, they sent me a letter saying that they were going to start charging a fee for foreign exchange transactions, rather than building into the currency conversion, as they had been in the past.
--
It's wierdo, not weirdo. Yes, I know that's not the 'proper' spelling of the similar english language word. ;)

Re: VOIP.MS Bank Credit Card Transaction Fees

Mon, 06 Oct 2008 17:21:39 EDT

MartinM :

said by PX Eliezer :

TO MARTIN at VOIP.MS: If this is an increasing problem, you folks need to alert people, and/or change to a USA credit card procesor for your USA clients.

I don't think this is an increasing problem so far we've add 2 reports about this. I'm unsure of the quantity of customers that didn't advise us however changing provider would not prevent this from happening, in this case it's the credit card issuer that is charging the fee, and not the company that process it (In our case, Internet Secure). Perhaps adding a notice to our website that your financial institution could possibly charge any additional fee for a transaction from Canada is a good idea.

Also, Johnnygood recommendation is a good one. You can add a credit card to your paypal account and pay via paypal to avoid this.

--

About incoming call issues, we didn't have any issues that I am aware of so I'll suggest contacting support about it.

Re: [Other] VOIP.MS

Sat, 04 Oct 2008 07:33:59 EDT

hockeynomad : You can choose the value or premium route can also be selected per call by dialling a prefix.

What is that prefix?

Re: [Other] VOIP.MS

Sat, 04 Oct 2008 00:22:18 EDT

gg22 : N/A

Re: [Other] VOIP.MS

Fri, 03 Oct 2008 21:19:48 EDT

hockeynomad : Montreal server or Toronto?

Re: [Other] VOIP.MS

Fri, 03 Oct 2008 19:54:39 EDT

PX Eliezer : I'm getting inbound on the New York server.

But being the New York server, "Thirty little birds sitting on a curb" comes through as "Toity little boids sittin' on a coib"

Re: [Other] VOIP.MS

Fri, 03 Oct 2008 19:28:08 EDT

Test99 : I'm getting incoming calls, using the Los Angeles server.

Re: [Other] VOIP.MS

Fri, 03 Oct 2008 19:20:59 EDT

gg22 : N/A

Re: [Other] VOIP.MS

Fri, 03 Oct 2008 19:07:15 EDT

hockeynomad : SIP URI forwarding?

Please explain in layman language.

Re: VOIP.MS Bank Credit Card Transaction Fees

Fri, 03 Oct 2008 16:15:28 EDT

JohnnyBeGood : Thanks for the links!
Seems like Paypal is way to go, at least for me.

Re: VOIP.MS Bank Credit Card Transaction Fees

Fri, 03 Oct 2008 14:31:32 EDT

PX Eliezer : It looks like your bank charged you a fee of 3 percent.
($ 25.00 x 3 percent = $ 0.75)

Here is a recent column by a travel expert telling about these fees:

"....(Customer): I booked a reservation and bought a ticket on Air Canada's website, using my regular Visa card. The ticket was priced in U.S. dollars, so when I got my card statement, I was surprised to find a "foreign transaction fee" of 3 percent tacked on to my charge. This happened even though the flight was codeshared and actually operated by United and even though Air Canada labeled the site I used as its 'U.S. site.' I couldn't find anything on Air Canada's website or the Visa site which makes any of this clear. When I contacted Visa, the representative said this fee is applied even if the purchase is paid in U.S. dollars with no currency conversion involved. I can understand a fee for Internet purchase for goods shipped into the U.S. from a foreign country but I would be surprised to learn that major airlines and hotel chains don't have any bank accounts in the U.S. and suspect that no actual currency conversion is taking place! It feels like one more nickel-and-diming bedevilment for the weary traveler......"

Link:
»www.smartertravel.com/travel-adv···=2663605

TO MARTIN at VOIP.MS: If this is an increasing problem, you folks need to alert people, and/or change to a USA credit card procesor for your USA clients.

ALSO:
Here is a link to a list of banks that tell you how much they charge for this fee. CapitalOne charges NO fee at present:

»www.creditcards.com/credit-card-···php#fees