Archives : November-2015
Home » 2015 (paged 8)
everyone We actually had alike problem: – user of a system can have multiple SIP accounts (to be able to receive calls on several devices, I suppose he cannot use only one SIP account, because theres only one peer registered, in this case) – to c..
All,Not sure if this is the right mailing list since the dahdi-dev seems not really active, so Ill try here.Im developing a new DAHDI driver for a custom board. In this card Ive implemented the reading of the TDM slots by 2 DMA channels, TX and RX. E..
,I have made a fairly complex dialplan where I am using the REGEX function in many places. This works so far, but I wasnt able to solve the following problem. What I would like to do is the following (please note that this is normal regex syntax ..
,Were trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doe..
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..