said by jeffnyc :

Another question, for those of you with the MG adapter, when you dial 0000, do you have to enter your phone number or does the voicemail system automatically know the number you are calling from?

said by hwittenb :

Assuming the PAP2T is like the PAP2v1 then you would have couple of elements like this:
|<00:1404>[1-9]xxxxxx|<0000:1404>[1-9]xxxxxx|
Skip the 01. part that is showing. I don't know how to get rid of it.

It's easy with just 00 but if you also can dial 0000 it is a little more complicated so that the 00 element doesn't preclude the 0000 element.

This is a good forum thread for Sipura dial plan basics:
»forum.voxilla.com/linksys-sipura···511.html

said by jeffnyc :

Anyone know how to alter the dialplan on the pap2t so that 00 or 0000 dials 14040000000 ?

said by sumdum :

i use their MG-3 if that matters.

it listed in my call history as;

>>>8/11/2007 12:45:32 AM USA-Georgia 14040000000

said by sumdum :

have you tried 0000, it has been working since yesterday.
you have to enter your 11-digit mailbox number.

said by jeffnyc :

Anyone else having issues with accessing voicemail? It ignores the press of the * key.

said by Cornhusker :

They are still Outbound Proxy Server, isn't it?

said by mazilo :

Does this mean VS no longer support the use of an Outbound Proxy server?

said by Cornhusker :

Server: 206.165.50.116 Port: 80

Server: 206.165.50.117 Port: 5060

said by rybshik :

I am having trouble to setup my Grandstream GXP-2000 with Voicestick.

I can receive incoming calls (including with G.729 codec). But can not make out-going calls. For example, when I dial 0000 to access my voicestick voice mail, I get the following error message:
488 not acceptable
try a different vocoder

said by DawnTreader :

I have gotten fax to work through VS...the most success I have had however is using my POTS line and using my VS number on bridge mode set up like a calling card on my fax modem. I have not tried with a stand alone fax machine.

said by jeffnyc :

Anyone get fax to work through Voicestick?

said by mazilo :

I just noticed the above $PING string. On a PAP2v1, $NOTIFY is the default string. Has anyone tried this $PING string on a PAP2v1 besides the OP and how is it different from the default string? Does it send out less packets?

said by w8sdz :

NAT Keep Alive Msg: $PING

said by pcmeet :

My VoiceStick has been acting in the last 4 days, drop incoming calls after 30 or 33 seconds and many times no audio, I am sure VoiceStick have their hands full with issues, Bruce can you let us know what is going on ?

said by nicklinus :

You can use Asterisk/Trixbox but their customer service will not help you if you have a problem. They will help you the best they can but they don't offer full technical support.

said by DawnTreader :

I do find it interesting that you were able to dial out. Did you get dial tone? Did is actually register? One thing I learned is that you have to have your number in the caller ID line not your name which I noticed you had. Best of luck and if you do find a way to do it let us know.

said by memorito :

I think you are confusing the PAP2 and PAP2 v2. The PAP2-NA is based on the PAP2 (v1)...

said by DawnTreader :

If you unlocked with CYT you may need to now flash it to the latest PAP2-NA firmware. This will keep it from dialing home to Vonage.

said by DawnTreader :

...I am wondering if your device is completely unlocked and the fact that you don't have access to all of the config settings also makes me think that this is the case.

said by pcmeet :

Any one knows how can I access this overseas router from USA, should I ask my friend to put IP address 10.0.0.2 in the DMZ and type his WAN IP address that is assigned to the router WAN?

said by pcmeet :

Any one knows how can I access this overseas router from USA, should I ask my friend to put IP address 10.0.0.2 in the DMZ and type his WAN IP address that is assigned to the router WAN?

said by ljjjjj :

Hi, I also have a Zoom ATA and have the same settings except for "Domain/Realm:" is set to "206.165.50.116". Also I do have the "Authorization ID:" setup. Hope this helps..

said by ljjjjj :

Hi, I also have a Zoom ATA and have the same settings except for "Domain/Realm:" is set to "206.165.50.116". Also I do have the "Authorization ID:" setup. Hope this helps..

said by goodchefro :

Did you read by any chance the VOIP and Phone Troubleshooting section and tried first to use any of the tips provided?

said by goodchefro :

Any chance someone uses this SJ L Softphone with VS and is willing to post the settings?...Thanks

said by BruceN :

I am not sure why it is set to this interval, but I think it is hard coded into our prior ATA so it may not be an option to do any other seting. Just a guess, not hard fact.

said by CyberSultan :

I have my proxy setting to i2telecom.com and the SIP proxy to 206.165.50.116. These configuration settings came from this forum. Are you saying this is not correct? You said two opposite things in the same sentence. I am using them without any problems. :)

said by CyberSultan :

I have my proxy setting to i2telecom.com and the SIP proxy to 206.165.50.116. These configuration settings came from this forum. Are you saying this is not correct? You said two opposite things in the same sentence. I am using them without any problems. :)

said by pcmeet :

No, I think it is clear, he has the wrong SIP proxy setting, SIP proxy should show i2telecom.com not 206.165.50.116 he needs to find out where he can set his outbound proxy and enter the value 206.165.50.116

said by hfrisch :

It is not clear why you are getting this error.

said by mickel1138 :

Well I got pretty far in getting my iAN-02EX to work, too bad since it's a really cool ATA with lots of features.

I still can't receive incoming calls, I can send the syslog to anyone interested. Incomming rings on the caller's phone, but no ring-out on the connected phone.

said by pcmeet :

Bruce,

Any way I can use message waiting indicator with VoiceStick?

said by CyberSultan :

said by kalimark :

Anyway, most of incoming traffic at home comes to our other numbers and the VS gets only the occasional return calls from Caller ID.

said by BruceN :

I am looking at a SIP number. So many of our customers do not use inbound, they are on a soft-phone and making international calls. We could cut costs and maybe lower rates.

Kind of like Skype out. Call out on to PSTN or SIP and incoming is SIP only.

said by goodchefro :

I hope I didn't antagonize anyone (especially Bruce and his boss :) ).

The idea I was trying to convey is that you should cater to both business-oriented (abusers)and "residential"/commercial users.
While to me it appeared that those folks using over 10,000 minutes per month. . .

said by pcmeet :

said by BruceN :

I am not sure if an OPT out is legal on the e911.

said by BruceN :

I am not sure if an OPT out is legal on the e911.
Bruce

said by Automate :

... Then, for every payed minute of outgoing calls give them 2 or 5 minutes of extra free incoming minutes...

said by mazilo :

I thought incoming minutes should benefit VS. Wasn't how those futurephone and/or FreeCallsToChina IOWA made $$$?

said by goodchefro :

I understand someone trying to limit its business expenditures, but talking almost 9 hrs/day continuously on the phone suggests a developed business.

said by goodchefro :

It the end it comes down to whatever the company envisions for its own good.

I personally think that "abusers" shouldn't dictate the tone of a business in terms of rates/pricing.
Set CONDITIONS/rules in place, but don't "punish" all of the other customers. Like pcmeet said, you can limit the amount of free incoming minutes, after which a charge should be inforced. If you abuse, you have to pay.
I understand someone trying to limit its business expenditures, but talking almost 9 hrs/day continuously on the phone suggests a developed business.

Leave the next2Nothing plan uncharged monthly, unless a customer goes over let's say 1,000 minutes worth of free incoming. After the free 1,000 minutes, a 1 - 1.5 - 2 cents/min (whatever you, the company sees fit for its own interest) should be charged. On top of that, overages should subject that particular account to a "monthly rate".
The monthly rate should be based on the level of minutes used beyond the 1,000 free ones, something like:
1,001 - 5,000 minutes -----> $ 1.99
5,001 - 8,000 minutes -----> $ 2.99
8,001 - 10,000 minutes -----> $ 3.99
10,001 + -------------------> $ 4.99.
I just made up these figures, but hopefully the general idea transpires.

I cannot use the service anyway, but I don't think you should change drastically your rates just b/c of some abusers. What you have to do is adjust your rating policy to abusing situations.

said by jinjimbob :

Once there is a monthly fee, people will expect a higher level of reliability with redundancy/fall over.

I like the low international call rates.

said by pcmeet :

You may set the rule for incoming calls to stop the abuse, I rent many DIDs that each comes standard with free 1st 5000 minutes and 1c per minute for over 5000.

Next2Nothing, you may offer the DID as an option and charge a fee ($2-$3) but at the same time US per minute should match France and UK at 1c.

said by BruceN :

Probably the reverse would be better, it would stop the abuse factor.

Forgetting e911, termination losses,and what ever, abuse by commercial users is the #1 problem. You ought to see the list! Some users doing 15,000 min. plus a month! Not a residential user for sure.

said by mazilo :

This is really a bad option to have. :(

said by jinjimbob :

Charge 0.1 cents a minute for all incoming, until a user spends $5 per month.

said by N9MD :

Hold on folks. Does anyone remember BruceN's original posts from 2005 when VS was in Beta. He was lambasted, parboiled, and other wise abused for suggesting that the first minute of each incoming call would be free --- and additional minutes would be charged. He cowered, simpered, expressed contrition and retreated rather quickly.

The argument against charging for incoming calls is that one is forced to pay for wrong numbers, telephone marketers, political campaign fundraising and the like. This would be particularly egregious for working folks who might come home to find scores of junk calls on their answering machines (or in voicemail), each of which was charged to their VOIP account.

said by jinjimbob :

I agree, make the e911 an option that would cost a monthly fee, most of us wouldn't choose it.
Charge 0.1 cents a minute for all incoming, until a user spends $5 per month.
I just ordered a Linksys adapter to start using Voicestick.

said by pcmeet :

I disagree with charging for incoming calls and it will not be in Voicestick interest because this will cause many customers to cancel and join other providers who don't charge for incoming minutes, as far as I know Voicestick pay for the DID but they should be making money on received calls otherwise how IPKALL stays in Business and why they cancel your DID when you don't receive calls for a month?

I do agree to make e911 an option with monthly fee.

said by voiplover :

Where did the "VoiceStick BYOD configurations needed" thread go?

said by slow mo :

Since I use VS I might as well chime in.

I agree with making e911 an option w/ monthly fee. For incoming calls it can be calls over, say 200 minutes/month will cost 0.1 cents/min.

said by N9MD :

The argument against charging for incoming calls is that one is forced to pay for wrong numbers, telephone marketers, political campaign fundraising and the like. This would be particularly egregious for working folks who might come home to find scores of junk calls on their answering machines (or in voicemail), each of which was charged to their VOIP account.

said by jinjimbob :

I agree, make the e911 an option that would cost a monthly fee, most of us wouldn't choose it.
Charge 0.1 cents a minute for all incoming, until a user spends $5 per month.
I just ordered a Linksys adapter to start using Voicestick.

said by slo mo :

I agree with making e911 an option w/ monthly fee. For incoming calls it can be calls over, say 200 minutes/month will cost 0.1 cents/min.

said by N9MD :

People! Stop looking at "unique products" and "i2bridge" offerings--- you are viewing this from the consumer's point of view. Put yourself in BruceN's place (the CEO of VoiceStick). It's all well and good (even, warm and fuzzy) to offer the customer a gold plated Bentley for the price of a Tin Lizzie --- but in the end, the provider will run out of gas, the company goes to the junkyard, and the customer is left to hoof it. A "loss-leader" is OK with companies that can draw you in with a lowball price --- and then sell you lots of add-ons to make the profit. The outgoing / callforwarding / bridging per minute charges (add-ons) will likely not amount to a gold-strike for VS, considering that many N2N VS users are "freeloaders".

I am not looking to be negative. I am trying to be pragmatic, realistic and sensible --- 'cause if the provider doesn't make a profit, the consumer will ultimately be out of business, too.

My suggestion is for company's like VS to stress the positive aspects of "pay by the minute" for outgoing calls while offering desirable features (voicemail, call forwarding, failover, CID, call blocking, etc.) --- rather than asking the potential customer to pay a fixed monthly amount for a set number of minutes (which may never be used by the customer). A $3 per month DID charge instead of the N2N Plan (which is what GizmoPlan charges, without a strong feature set) might just generate a financial cushion for VS while being seen as reasonable for the offered features by the end user.

said by global_dev :

that is probably the single unique product offering VS has that other companies do not (other than i2bridge, which many other companies will probably be implementing soon).

If they change that, they may as well be any other company that charges for a DID.

--- but you may be giving away too much with the N2N Plan (and from your previous comments, it may be costing you a pretty penny)

said by pcmeet :

...Another way to re do the VS400 is to change it to VS500 and stays at $9.99.

said by BruceN :

... the VS400...We are going to re do plans. If anyone has ideas as to what you think would be a good plan offer, let me know.

said by CyberSultan :

I have found the VS sound quality to be decent, but VS currently lacks some of the great features that other providers have. Bruce has stated that they are going to be re-doing the systems and adding features. Not sure what the exact ETA on that is, though. Would be great to have that ETA, as well as a preview of what features they expect to be adding to the service. Perhaps the system upgrades will solve some of the problems you are experiencing. I know that doesn't help you right now, though. :huh:

said by goodchefro :

Beats me...
Bruce tried to help me over last weekend with the softphone, but no luck.

said by goodchefro :

It is beyond interesting for me, because Stanaphone and CallWithUs have no problems whatsoever and the voice is great.
I also use neither DMZ, nor port forwarding.

said by goodchefro :

What's going on here?

said by goodchefro :

At any rate, it might be possible that the softphone interferes with the ATA? Although I never had them both operational the same time.

said by goodchefro :

Could it be that VS in particular is not compatible with connections below 768 Kbps

said by goodchefro :

I changed the codec, but no improvement in voice quality. :(

said by goodchefro :

- When accessing the web-based utility the date always changes to some date in 2003.

- If STUN is enabled, should the port 3478 be made available? Anything needs to be entered in the "STUN Server" tab?

- I cannot dial 0000 for Voicemail. There is a blank "Voice Mail server" box in the Line1 configuration tab, but I don't know what to enter. Is that the matter or something needs to be changed in the dial plan?

- Most important problem is that voice quality is choppy, inbound and outbound.

I have a generic Linksys cable/dsl 4-port router. I entered the port range (5060-5061) in router's Port Triggering tab, but have no idea if that makes a difference or not.

said by goodchefro :

I have a PAP2T-NA configured with VS on line1, and updated with the latest firmware 5.1.3(LS).

said by Trigolis :

Just tried that and it didn't work. My firmware version is 3.1.18(SE). What does the 'c' stand for in your version 3.1.8(SEc)?

Elapsed Time:1 day and 05:00:40

said by Kill4Speed :

Trigolis,

Try toggling the "Line Enable" (line1)
What firmware version are you running your SPA1001 with?
Mine was 3.1.8(SEc)
I do not know if it was just coincidental, but toggling the said settings fixed mine (line2) registration problem

v,

said by smartbiz :

Hi!!!!

I am using SPA-1001 with Line -1 Configured for Voipcheap & Line -2 to Voicestick at my work as well as home.

I would suggest do a factory reset on your SPA and reprogram it.

If you want a screen shot then email me

said by slow mo :

Well, your SPA-1001 line 2 may be defective. Switch Voipbuster to line 2 and Voicestick to line 1 and see what happens.

said by Trigolis :

My VS account is active. I just tried the softphone app and it called out fine.
:(

said by slow mo :

Are you sure your VS account is active? Login on the web to see. If yes then you should use Voicestick's softphone and see if it's working.

I've tried the setting on my PAP2 (similar to SPA-1001) and 1402S and both work just fine.

said by slow mo :

I think you should now enable stun server. Keep everything you have now and Use Auth ID should still be no.

I use 217.10.79.2:10000 (stun.sipgate.net:10000) on my PAP2.

said by Trigolis :

Did that but still getting failed registrations state.

said by slow mo :

Subscriber Information;

Use Auth ID: change to no

The rest look good.

said by Trigolis :

I'm having trouble getting my Sipura 1001 to work with Voicestick. I followed the settings at this link: »www.voipconfig.com/voicestick_config.htm but I'm still getting a registration failed status.

I have Line 1 configured for outgoing with Voipbuster and this works fine. Getting Line 2 to work with Voicestick is problematic. I've checked and triple checked all the settings, logon ID, and password.
I would like to take a snapshot of my settings and paste it here but not sure how to go about it. Anyone have these problems before?

Also, I'm planning on getting a DID that has SIP forwarding. Is it possible to forward all SIP calls into my Voicestick account?

thanks

said by carmatt59 :

After you write the body of the post, you can click Previw/Attach, where you will see an " + upload attachments". You can browse for the file you want to upload, then just hit upload.
Hope this helps.

said by Trigolis :


I would like to take a snapshot of my settings and paste it here but not sure how to go about it.

said by flroots :

Do I understand correctly that when calling out with voicestick you do not hear the ringing in the phone's receiver ? I wish to know if this is a problem unique to me or universal. Thanks.

said by flroots :

With the help received, I believe I have a good voicestick setup for my Grandsteam HT-496.

said by hfrisch :

Some times I hear the ringing tone feedback - sometimes not. It is inconsistent - but it is easy enough to get used to. The calls still go through properly.

said by DSLrgm :

I cannot find any reference to BYOD on the VoiceStick web site.

Why?

Also, I would be happy to work with you on a 'safe' Asterisk config. If I were to apply myself, I know I could get Asterisk to look like some of the ATAs and/or softphones you are accepting.

Also what about Linksys WRT54Gs with firmware that makes it a SIP server?

said by voiplover :

Little off topic but;
Can you please post the link to the SIP server firmware for the WRT54G that does that.

said by BruceN :

Any of you ever used Open Wengo ?

said by bakermd :

tkmoult: Thanks for the tip. We are working on a network project to migrate away from some legacy products we are using, and as soon as this migration is complete, we will be implementing ENUM support. SipBroker seems to be one of the larger databases, and so I will test this one with our switches.

said by flroots :

Does anyone have a Grandstream configuration working with VoiceStick? I have the Handytone HT-496, but suspect that any Grandstream configuration would be helpful. Thanks.

said by voiplover :

My guess is that most VS subscribers would mostly look towards using a soft phone on their laptops while they are on the move.
What do you think the ratio of Linux vrs Windows operating systems is on these laptops??? :uhh:

said by N9MD :

From what I see on the Twinkle and Kphone sites, they are softphones for use only on a Linux setup. That leaves a "few" of us out in the cold. ;)

said by voiplover :

My guess is that most VS subscribers would mostly look towards using a soft phone on their laptops while they are on the move.
What do you think the ratio of Linux vrs Windows operating systems is on these laptops??? :uhh:

said by N9MD :

From what I see on the Twinkle and Kphone sites, they are softphones for use only on a Linux setup. That leaves a "few" of us out in the cold. ;)

said by sokhapkin :

Twinkle and Kphone work fine with VS. However, I will stay away from VS because of annoying VS SIP server misbehavior, the server ANSWERS the call immediately even if the call can not be completed, that can lead to billing of unanswered calls!

The SIP flow is:
client->VS: INVITE
VS->client: TRYING
VS->client: OK -- the call is answered, timer starts!
VS->client: (in audio stream) long rings.

The right behavior is to send "183 Session progress" to the client (or BUSY). SIP OK should be send only when the called party answer.

said by sokhapkin :

Twinkle and Kphone work fine with VS.

said by BruceN :

...Anyone here in this thread ever need a favor, just ask!

said by slow mo :

You will always get a busy signal when dial your own number. That's how telephone works. :)

If you want to dial 0000, just add it to the dial plan:

(*xx|0000|{:1aaa}[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|{:1}[2-9]xx[2-9]xxxxxxS0|011xx.)

aaa - area code

Replace { and } with (less than symbol) and (more than symbol).

Normaly from a phone that has voicemail I've been able to dial my own number and it sends me right into voicemail.

I added 0000 to the dialing plan and am able to connect into my voice mail box. It doesn't ask me for a password It then ignore any options I press from my phone. I did call into my cell phone and that provider does work with my touchtones. I'll have to just call in from my other line to setup the voicemail.

Also it is not sending my callier ID info (shows as unavailable number) when I call out. is this normal?
EDIT: I don't use voice mail on Voicestick so I never put in 0 or 00 or 0000.

said by slow mo :

I have a working Voicestick on my PAP2. I figure they are very similar. So this is the setup.

said by BruceN :

markosjal....

Not sure why you did not get an answer to your question about rates. We double check every rate, but I would still bet that there are a couple errors in the file. Can you tell me which ones they are? I will be happy to look at them, and fix and credit if broken.

Bruce i2telecom

said by sokhapkin :

Wrong assumption. VOIP providers have to purchase DIDs to offer to their customers.

said by digiblur :

I've always thought the incoming calls didn't cost VoIP providers anything.

said by BruceN :

The free incoming number with voicemail is a great overall cost for us.

We pay for the DID number, E911 to Intrado, and of course the incoming calls we pay for.

said by christcorp :

The dialplan should have nothing to do with getting a dial tone instead of a fast busy. A fast busy means that it can't find the SIP server that you are wanting to use, or that it's not configured properly for a call.

said by biomesh :

Does your device show line 2 as being registered?

If it does, then you might want to change your dialplan.

You should also clear out the "Auth ID:" field as it is not needed.

said by MannyL :

I'm trying to setup an unlocked RT31P2 with Voicestick on line 2 (Line one works with current carrier) When I pick up the phone on line 2 I gte a fast busy signal. would someone review my settings and tell me where I went wrong please

said by gimp55 :

Hi Bruce,
I forgot to mention if i leave the SIPRegInterval at 3600 it times out and no one can call in, i have to reset the ata, but setting it to 600 works great.
the jpg i posted has it at 3600, i need to edit that.

said by PhilAIV :

Yeah, you cleared out my number too, lol. I didn't use the number anyway, so you deserve to put it back in the pool, but I noticed that you don't offer any more 212 numbers. Ran out or too expensive since they are heavily sought after.

PhilAIV

said by BruceN :

gimp55
I am doing a cleanup of dead accounts and hope to have a ton of DID's up in about 10 days.

said by Quattrohead :

Is this really the correct price for UK standard calls with next to nothing plan ? 0.0090

said by N9MD :

Bruce --- you are only mildly neurotic with 10 numbers and 12 steps in Voipaholics Anonymous.

My 64 current numbers from 10 VOIP providers (using ATA devices and/or SIP-URI softphones) led my wife to enroll me in the VOIP rehab program (72 steps) at the Betty Ford Clinic. Could that be because every time she calls friends and family, they won't answer because they don't recognize one of the myriad of numbers that show on Caller ID?!

ZAP!! Yow! That hurts! --- At the moment I am in the Electro-Convulsive Therapy (ECT) ward where I am required to recite my VOIP numbers one at a time, with each recitation being followed by a 1 million volt burst of electricity. (Fortunately, they are still using very low amperage.) :D :D :D

-=-=-=-

Meanwhile, as a VS user & non-abuser (only two numbers, one in NJ and one in FL) I like the idea of auto-deposits where the customer gets to select the amount.

said by gimp55 :

Same issue here, money in account, worked fine with ata186 till recently, incoming calls work fine.
I have to keep switching over to my sellvoip account to make outgoing calls, would rather use VS but cant.
here is what i got back from support, settings worked fine till lately:

Your ATA is your responsibility as we do not support this device since it is not a certified product on our network. Here is our SIP configuration for your ATA.. double check it as something might be incorrectly set up...

The information you need to provision your 3rd party IAD is the following:

SIP Proxy IP: 206.165.50.116
SIP Proxy Port: 5060
SIP Registration Domain: i2telecom.com
STUN: Disabled
SIP Username/Password: Your DID #/ your password

Be advised that we do not provide any troubleshooting or maintenance support for 3rd party IAD’s. Also, the provisioning information can change at any time.

The i2 Telecom Customer Care Team

said by markosjal :

Just my two cents about VoiceStick.

I tested VoiceStick as I have many other providers. I always need to know what the competition is up to!

My main complaint is that on International calls I often received no call progress tones, just silence till the line was answered! If the called number was busy, I would hear nothing!

I recently re-tested with the same results!

Also I had sent a message to support, through your web portal, noting that the billed amount of at least one destination was different than the posted rate. I believe it was Guadalajara Mexico 52 33 XXX XXXX . I received no reply to that message. Accurately billing according to the posted rate is a basic customer need (and I would say a right!). It is a shame it was not replied to.

I do not want to sound too harsh here, but I need to post what I feel is the truth.

said by jdl20 :

I have a cisco ata 186 configured for voicestick, I can't make outgoing calls because of this unnecessary reinvite. It seems like this reinvite thing is something that VS implemented a few weeks ago, because I used to be able to make outgoing calls. I can only call my voicemail at this point because there is no reinvite when I call voicemail.

If VS is serious about their BYOD plans, they should consider getting rid of this unecessary reinvite.

said by BruceN :

...BTW, I love your tag line. My wife says the same thing to me. I am a phone number aholic, in need of a 12 step program. I bet I have 10 numbers.

said by BruceN :

Well I guess this is back to my point. We do not have hardly any info on using other devices on our service. Any of you smart people have any thoughts?

said by BruceN :

I doubt that the reinvite is doing you in. We have made no system changes. I wonder about your account. No money in it? bad CC date, or some other issue.

This is most often the cause of this sort of problem.

Let me know if you want me to look at your account.

Bruce

said by sokhapkin :

3. Unneeded REINVITE on outbound calls. When the outgoing call is placed, your server responds with
SIP/2.0 200 OK
c=IN IP4 206.165.50.116
t=0 0
m=audio 35898 RTP/AVP 0 101

Session progress audio use this RTP port. But when the call is connected, your server does REINVITE and wants different RTP port:

INVITE
c=IN IP4 206.165.50.116
t=0 0
m=audio 35900 RTP/AVP 0

What is the reason for reinvite if the audio server IP address did not change? Moreover, some SIP clients have problems with reinvite. I can understand REINVITE to forward the voice path to another server in order to not run the traffic through your SIP server, but see no any reason in REINVITE to the same server.

said by BruceN :

How far is Liburn from Roswell?I am a newbie!

100% of the time when we have a customer signing up for a free account AND it gets rejected we see a 100% fix of the problem by having the customer select the $5 button (And we will void that after the fact if desired)

said by BruceN :

I am not looking to pick a fight with Mazilo, but I am wanting to give him some of my thoughts, and not hide behind the curtain.

said by BruceN :

But do bear in mind that we are interested in learning from this group. Just because we run a company does not make us right!

Thanks for the input, and I will contact Mark about all this.

said by BruceN :

So, I think overall, we are on the right track to get customers great service.

Will it be perfect, heck no, but it is the goal.

said by BruceN :

And hey.... I am one of the key people in the company and I am silly enough to post my email up here.

said by w8sdz :

Most web sites that accept credit cards check the address you specified during your sign-up against the mailing address the credit card company uses when they send your monthly statement to you. If these addresses do not match the transaction is denied.

said by mazilo :

It looks like his CC is no good with VS auto checking systems that every time he wanted to open for a new account the system responded with the following error message...

said by mazilo :

•It looks like his CC is no good with VS auto checking systems that every time he wanted to open for a new account the system responded with the following error message...

said by w8sdz :

VoiceStick will soon be adding BYOD support. They would like users who have successfully configured ATAs for their service to post their configurations here. Both Softphone and hardware devices are welcome, however Asterisk will not be supported at this time.