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…