Archives : April-2014
Try starting Asterisk with the -f option.It will NOT fork into the background so you will see all messages on startup (including any that might not end up in the log file).Search for DAHDI errors which will likely be there.Also, if you configure everyth..
Its my first post here, so Ill cut to the chaseI have 2 Asterisk servers and want to connect them using sip on one and pjsip on the other one. One is running at home and another at a VPS. The first one will be the client (with dynamic ip) and the ..
We cant do much with part of your debug. Youll want to post a pastebin link to your full SIP trace, and be sure that it includes at least VERBOSE messages turned up to 5.[1]Work on WebRTC support is on-going, so youll want to test in the very lat..
From the reading and testing I have done it doesnt look like SIP supports a username and password in the Dial string. I currently have the following mapping.priv => dundi-extens,0,SIP, dundi:pass@1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartia..
Guys, Just new to Asterisk and am completely stumped.I have created two accounts as instructed.Please see below for the config of the user accounts. [Peter]type=friendhost=IP addressdisallow=allallow=ulawallow=alawcallerid=Peter secret=XXXX..
I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release:1) Asterisk 1.4.21.22) Libpri-1.4.43) Zaptel-1.4.11I would like to move the OS to CentOS and then I thou..
I asked a similar question yesterday but unfortunately I somehow got disconnected from this group and I may have missed a response. Wanting a systemd start script for dahdi for archlinux. The install does not seem to make one. Has anyone created ..
Ive had years of experience using ODBC for CDR, SIP, and extensions with Asterisk.One thing that has been problematic in the past is with documentation as far as database tables changing between versions (even within minor releases, though that was b..
On 11.9, trying to set up a callcentric peer:sip debug:asterisk is trying to find a peer based on the _calling number_!Heres the callcentric peer based on its support pages:[callcentric]type=peer context=from-callce..
I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts.Firefox:Neither sipml5 or jssip works with calls to asterisk, audio/vi..