similar to: first dtmf is not detected

Hi, I have encountered a DTMF issue. My scenario: Access carrier-----sip----> Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk forwards it with SIP INFO method to Cisco gateway, but on TDM switch every digit is duplicated. Is it possible that the carrier sends inband along with rfc2833? Kind

Hi! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Thanks a lot, Szasz Szabolcs

Hi, How can I strip + from the front of the caller ID? I have tried this: exten => s/_+X.,1,Set(CALLERID(name)=${CALLERID(name):1}) But it is not working. Szasz Szabolcs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100114/a3f18781/attachment.htm

Hi I registered my asterisk box to my SIP provider as an UA. For every call I receive on this trunk, I get the message "That is not a valid conference number". I'm using Asterisk version 1.4.22, I had install the dahdi-linux and dahdi-tools and the conference is working between the phones registered to Asterisk PBX. What's wrong? Thanks. Szasz Szabolcs -------------- next part

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

Hi, How can I disable comfort noise on Asterisk? Szabolcs Szasz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100129/3b27a653/attachment.htm

Hi all, I have a lot of call in wich I found that my Asterisk doesn't answer the BYE message, then the BYEs are retransmitted, but the call ends, when the Asterisk sends a BYE. Time AS.TE.RI.SK CA.RR.IE.R1 0 INVITE SDP ( g729 g711A g711U telephone-event) SIP From: sip:1265666072 at 81.209.186.14 <sip%3A1265666072 at 81.209.186.14>To:sip:1234567890 at CA.RR.IE.R1 (5060)

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > >

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

Dear list members, Even though I found extremely reasonable not sending any audio when a PROGRESS message is received on a PRI channel (isn't it an early-media session or one-way audio session?), nevertheless some Italian IVRs expect the user to select the proper option by sending DTMF. Now my asterisk box understands correctly the DTMFs on the caller SIP channel, but it doesn't forward

Hello Does someones know a good low-level way to debug the DTMF that is arriving (or not arriving) at Asterisk? We've already used the DTMF logger level. The picture is the following: We are developing an application that calls to a customer to get authorization information through DTMF. We've used at development environment SIP/IAX channels and everything was great and the work was

Hello, I did this post a long time ago but never solved the problem, so i'm trying again after something like 10 months, hopefully i'll find someone that found a solution ;-) When i call an external number that sends audio before call has been answered (like some PBX of public offices do here in italy), strange things happen: I'm using chan_capi, with Early B3 active, i can listen

Hi, I need some help, getting to work asterisk MWI. I set up Asterisk as voicemail server for Openser as this tutorial shows : http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 . My voicemail system is working but, I can't get to work the message waiting indicator. It doesn't seems to send the Asterisk any NOTIFY message to the Openser box. How can I

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other users. Other DTMF lost somewhere. We have tested only with sip phones. Can someone help me with this, or

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other users. Other DTMF lost somewhere. We have tested only with sip phones. Can someone help me with this, or

Hi, I've noticed that asterisk seems to stop relaying DTMFs after a call has been up for a while (~10 mins). I was just wondering whether this was intentional, or a bug. In detail here's my setup SIP Gateway --> Asterisk --> E1 --> Asterisk --> SIP Gateway The LHS gateway sends RFC2833 DTMF messages to the first Asterisk which bridges them onto the E1. They then get

Hello, I setup a door open system with a basic DTMF card. The card is connected to an Sipura/Linksys 3102 FXS port and is powered by this port. My problem is that when I send a call with Dial() command, channel has to be answered before receiving DTMFs, what my card does not. Is there a way to autoanswer those type of calls or to send DTMF on an non answered channel or another solution/idea

Hi All, would Asterisk 'transcode' H.245 alphanumeric DTMFs to an H.245 signal / rfc2833 H.323 device over G.729 codec ? Thanks for supporting, .TF