We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance.
I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility, but this hasn't helped at all. The provider has suggested RFC2833 out-of-band, but the Cisco manuals do not clearly state which setting this is on the handsets.
I have enabled DTMF logging and spoken to the SIP provider, but they…
I build a conference server using Asterisk 1.8 and the third party module app_konference.so. I would ask on their forum, but the forum seems to be pretty dead. The problem I am having is that when I have conferences that have a lot of members, say 100+ users, the DTMF seems to not work. For example, pressing the digit to mute all conference members does not do anything. When I have smaller conferences, say 25 members, everything works the way it should. I would greatly appreciate any input at all on this issue.
I'm attempting to find where my extra long DTMF Tones are coming from.
I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway.
I'm pressing 4 to select a menu and everything is fine.
[Feb 12 16:58:18] DTMF channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF channel.c: DTMF begin passthrough '4' on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF channel.c: DTMF end '4' received on SIP/trunk-0a02dee0, duration 150 ms [Feb 12 16:58:18] DTMF channel.c: DTMF end accepted with begin '4' on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF channel.c: DTMF end passthrough '4' on SIP/trunk-0a02dee0 [Feb…
Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you know is there is a plan to develop it? 2. When we setup 2 peers, one RFC4733 and the other inband, the asterisk does not transcode the DTMF signals, therefore DTMF is not working. It used to work on release 11. This is really…
I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) -> PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record audiofile1,Wait for a tone,Record audiofile2). A few points to keep in mind, (1)I can not send DTMF tones as Conference system suppresses it. (2)There is no other way to pass information from Asterisk1 to Asterisk2 (3)Asterisk2 doesn't know the length of audiofile1,audiofile2. (files are less than 200 Sec in duration)
I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0.
In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked.
The syntax for ParkAndAnnounce I used was this (I don't want any announcement to be played):
exten => s,n,ParkAndAnnounce(,3600,SIP/100)
In the new Asterisk-version, the ParkAndAnnounce application gets called, but the call isn't parked.
The only error I can see in the messages file is a DEBUG entry saying that the…
I am looking for advice about the design of a SIP-based intercom. I count on your help, as my current attempts are not fruitful (yet).
This will be a pretty long message, so here's my fundamental question:
Is there a way to interpret DTMF tones sent by the calee (not the caller) while a voice call is in progress?
Here's the desired scenario:
- there is a box with speakers and a mic - Asterisk is running on a computer inside that box - the box is embedded in a door - There are two user accounts, UserA and userB - UserA…
Trying to properly broadcast / relay DTMF digits to other confbridge users, but does not appear to work. Goal is to have a conference user be able to receive the DTMF, so it has the effect of being 'broadcasted.'
I have the following set up in 'confbridge.conf': dtmf_passthrough=yes
From logger.conf, I can see the DTMF tones via setting "console => dtmf". When I dial into the conference bridge with a SIP UA and dial 9, for example, this is what I see:
sip1*CLI> [Dec 19 01:29:50] DTMF[C-000005ba]: channel.c:4164 __ast_read: DTMF begin '9' received on SIP/3002-0000003d [Dec 19 01:29:50] DTMF[C-000005ba]: channel.c:4175 __ast_read: DTMF begin…
Upgrading an ancient customer installation... was running 184.108.40.206
(Trixbox) with Zaptel 220.127.116.11 and a Sangoma A102D, which has been running fine for 5+ years. Customer getting anxious about hardware failure, so we built a new box and installed 18.104.22.168, Dahdi 22.214.171.124, and a new Sangoma A104D. The single active span is an RBS T1
I tried to move one span over one night which was working fine on the old box. Once plugged in there were no alarms, Sangoma wanpipemon utility showed "connected". I tried calling in on a DID number, and in the 'full' log, with debug and…
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 is there any configuration option that can resolve this problem? I want asterisk receive the DTMFs send at the same time and to pass those either by queuing them or by…