Topic 'Re: [Other] VOIP.MS' in forum 'VOIP Tech Chat' - dslreports.com

Re: [Other] VOIP.MS

Sat, 18 Jul 2009 23:39:17 EDT

PX Eliezer posted : 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 posted : 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 posted :

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?

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

m. :)

Re: [Other] VOIP.MS

Fri, 17 Jul 2009 17:46:51 EDT

espaeth posted :

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 posted :

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 posted : 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 posted : 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 posted :

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 posted : 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 posted :

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 posted : 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 posted : 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 posted : 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 posted : 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 posted :

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

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 06:29:48 EDT

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

Re: [Other] VOIP.MS

Mon, 13 Jul 2009 06:15:44 EDT

Mango posted :

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

Re: [Other] VOIP.MS

Sun, 12 Jul 2009 19:54:26 EDT

wysiwyg1972 posted : 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 posted :

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 posted : 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 posted :

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 posted : 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 posted :

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 posted :

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 posted :

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 posted : 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 posted :

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 posted :

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 posted :

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 posted : 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 posted : 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 posted :

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 posted :

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 posted :

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

Re: [Other] VOIP.MS

Fri, 12 Jun 2009 18:02:11 EDT

WildChild posted : 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 posted :

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 posted :

said by WildChild:

Who is your current carrier?

Bell

Re: [Other] VOIP.MS

Thu, 11 Jun 2009 20:06:31 EDT

WildChild posted : 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 posted : 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 posted : 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 posted :

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....