We have a Genband C3 Switch and a couple of customers that operate asterisk PBXes connected via SIP Trunk. All of them still use some 1.6.X asterisk and this works fine.One customer uses a 1.8 version and has a very strange problem:Asterisk 188.8.131.52-1digium1~lu..
all, I attempted to make a couple of outbound calls this morning and always got the busy tone. I checked the Asterisk console and was greeted with: [Dec 29 11:29:22] WARNING: app_dial.c:2218 dial_exec_full: Unable to create channel of type I..
Hello We want to implement T.38 passthru with asterisk 1.8.5 [Asterisk 184.108.40.206 built by root @ asterisk1-8.labdomain.com on a x86_64 running Linux on 2011-08-26 21:31:22 UTC] The call flow is: quintum gateway –> asterisk –> Dialogic IMG 1010 the c..
all, I am usingcall files to dial out to a set of PSTN numbers. The calls are going out fine and being handled correctly by the dial plan. The problem occurs when I accidentally call a fax machine. I would expect the dial plan to pick this up and j..
Hey Guys! I have two asterisk 220.127.116.11 same version on both machine but why one asterisk has reload command but other doesnt ? satish-desktop*CLI> core show version Asterisk 18.104.22.168 built by root @ satish-desktop on a x86_64 running Linux on 2011-03..
, Someone may have run into this problem. Very strange. I have a customer running 1.422. They use a digium ISDN card connected to an primary rate for their inbound currently. We have tested inbound SIP from one of our trunks. We use these trunks w..
Hi! Did play around with queues and need some help. I thought that MEMBERINTERFACE and MEMBERNAME should be set to the “device” the call was queued to not the device that called the queue, or do i miss something? Running: Asterisk 22.214.171.124 built..
I am facing an audio-problem with the dial application and I (!) think, that it is connected to the dahdi parameter overlapdial=yes. Sangoma support does not see any connection between this. But when enabling this option I face with some(!) dial-partn..
Let me explain: When I dial into Asterisk ( I have a SIP trunk – which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing – the extension rings on the Asterisk server (you ..
gang, I have a fun one for you.Im not getting the quality of sound I want out of GSM, so Im trying to make my files into .WAV (.wav) format.Here is the file output for 5 files: file *.WAV cents.WAV:RIFF (little-endian) data, WAVE audio, Microsoft P..