Im remotely managing an asterisk setup using an OpenVPN client on this Asterisk box, connecting to an OpenVPN server of mine).This box is mainly connected to PSTN. It is also connected to the Internet, only for remote management.The former ADSL l..
I had an old Asterisk installation die recently and we decided to upgrade to Asterisk 13 to replace the old server.Everything seems to be working with PJSIP but there is one issue.Asterisk talks to a callmanager via a SIP trunk and send calls to P..
We have a FreePBX-12 / Asterisk-12 setup that supports about 24extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence.The server host is a dedicated atom(tm) box using the Free..
I have a client that wants a phone system that will play sounds from asleep machine. I tried using all different formats (GSM, WAV, WV49,MP3 etc.). Over SIP it was OK however with the PSTN it broke up fromtime to time. I assume this has to do with ..
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..
Ive got a small install with Asterisk 11. This box is connected to PSTN through a SIP trunk.I need to add a cellular phone as a remote agent of an existing queue.At the moment, this queue is configured according a ringall strategy and busy agent ..
There isa PSTN line shared between 2 asterisk servers, (openvox 4FXOlines)The outgoing call of the one server may be conflict with the established call of the other one,is any way to force the Asterisk or Dahdi to dial after hearing the D..
On connection to an incoming call via PSTN where asterisk [11.7.0] is Dialing an internal extension on answering the call there is about 6-7 seconds before audio is heard on either side.When looking at the CLI traces when I answer the incoming call t..
On a Asterisk 1.8.12 system working OK for months (>100k calls proceed), users are complaining for bad audio.My setup is:PSTNAsteriskSiemensHiPathPSTNasterisk -rx dahdi show versionDAHDI Version: SVN-trunk-r10414 Echo Canceller: HWECasterisk -rx ..
B.H. allIm using Asterisk 11.7, connected to PSTN using SIP trunk.Im looking for a way to get data from INVITEs SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the c..