similar to: VoicePulse DTMF Problems Anyone?

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing

I have an IVR application that works fine from multiple DID sources, unless the call to that DID was from a Vonage service user. In this case about half the DTMF tones never get recognized by Asterisk. Has anyone else seen this? Suggestions? I'm running 1.0.9.

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

I'm looking to set up multiple inbound Voicepulse Connect lines and have Asterisk route them direct to different IVR or Voicemail based on the inbound number that is called. Unfortunately, I just can't see how one would go about identifying the number that is being called. Has anyone been able to do something like this with Voicepulse? I appreciate any assistance. Phil

Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and

Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls,

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

I have working SIP numbers with broadvoice, and just added a DID from http://connect.voicepulse.com/ . The calls answer, but DTMF is not recognized. With "iax2 debug" active pressing DTMF does nothing. Zilch. Zero. A friend tried a different IAX2 connection, and got the same results. I see the following in the archives: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: > Hey

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Hello all, I am using voicepulse DID's to receive calls via IAX to and asterisk IVR dial plan I have put together. The problem is after 3-5mins the system cant pickup the DTMF tones I am sending... I have tried different telephones... It just repeats menu options over and over.... I have to call back and then it works again for another few mins... Any ideas... iax.conf? issue? Thanks, J.C.

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

I have 1.2.12.1 Voicepulse using IAX I get about 30-40% issues with not having the DTMF tones work. I have 3 questions #1. Voicepulse says they are sending them, Is there some setting I can adjust to make sure my end is working? #2. I have set the Dialplan to play a sound "Operator" then go to a SIP phone. I would like it to write Caller ID Time .... to a file I can read and find

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

I run an automated information retrieval system, using Asterisk. Fairly often the system misses a dialed digit. Our codes are all 4 digits, see lots of logs with: 4199 - OK 530 - Invalid code 330 - Invalid code 5330 - OK As callers experience skipped codes. We're using Broadvoice SIP with inband DTMF (and we've tried every possible setting or option

I've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request

I'm having a very annoying problem with access my asterisk system from work. Our phone system here only produces very very short DTMF tones. The phones work fine for other IVR systems (Dell Support, HP Support, etc, etc). However, tones to Asterisk just never make it. The way I'm calling into my Asterisk server is such: OFFICE PHONE => CALLUK.COM 0870 => IAX Inbound The

I can't seem to dial out with Voicepulse Open Access service using *. Incoming works fine. Another user posted a few weeks back that they were having problems and there are some threads at dslreports.com about this as well. Maybe someone here can figure out what the issue is from the sip debug info below. I am at a loss. The audible error message from Allison is 0984 (from VP server) Here is

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

Hi all, I'm experiencing a problem with meetme i can't resolve. This is my scenario: A iax client, say IaxComm, make a call through a zap channel. When it answers it is tranfered to a conference room. Then the iax client make a second call though a second zap channel, at the other side there is an IVR. Iax client send some dtmf to the IVR then it transfers the IVR to the previos