Archives : November-2015
Would like to inform community that there is a small demo lab that one can FREELY use for checking DTMF delivery and two-way speech. Also it demonstrates passive (non-intrusive) voice quality analysis based on PVQA.http://voxserv.ch/demolab.htmlF..
I am trying to forward number, in the past I was able to use this:;;;201-704-4482exten => 4695,1,Dial(dahdi/8/w73#w7044482)exten => 4695,2,Congestion exten => 4695,102,Congestionis that correct way to forward? the phone is with AT&T company. On A..
I have configure bridgeConference. But im having some issue. I want to give the ability to the user when dialing from the Conference to hangup the call by sending dtmf tones without being hangup from the Conference. For example if the user call s..
We are launching a new product to help-us to reduce mobile call costs using Asterisk. More informations you can see at http://asteriskdialer.com.br/en — Att, Hélvio Junior SafeId – Gestão de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com..
Were almost there everyone!The OpenSIPS Summit in Austin is 1 week away. Theres still some room left for all you procrastinators!We also have a few spots left for the OpenSIPS LIVE Bootcamp following the summit. Enjoy a rare opportunity to train with..
The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP.There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use aster..
Im not getting any ringing when I use option r with Dial:Dial(DAHDI/1-1, motif/8447/+1@voice.google.com,,rTt) in new stackOtherwise all works. The call goes through, good ..