Has anyone used Asterisk for a Call Center operation? What I mean is: given a list of phone numbers, can Asterisk dial each number, play a message and accept some DTMF? I ask because I am an employee of a non-profit company based in San Diego, CA. I already evaluated Voicent and Voxeo. The former has expensive licensing terms and the latter is not best suited for a call center. I would appreciate your kind comments.
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…