Archives : October-2018
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i have webrtc client chrome69/jssip which is connecting to asterisk 13.23.1/pjsip i have strange problem where pjsip aor stays in status created sip trace on asterisk looks ok. do you think if this can be bug? test*CLI> pjsip show aors A..
Im testing an Asterisk instance. At the moment, Im focusing on its capability to receive and challenge incoming SIP Registrations.For various reasons, I would prefer to use SIPp instead of Asterisk to act as SIP Client.Has someone successfully done t..
Asterisk 16.0, PJSIPFor the first caller to a conference, I want to dial out and bridge that conference to a new PJSIP external call.For the next callers, I just want them to join the local Asterisk conference.After the last caller leaves the conferen..
I am currently evaluating asterisk 16. I have noticed an issue using application playback. The beginning and the end of the audio file are missing. If I use answer and wait(1) before playback, the beginning is correct. I am using chan_sip, if this..
Hi. I have three servers running corosync and pacemaker, to maintain a floating address between them.This is working fine, and I can, for example, SSH to the floating address and get to whichever server has the address at the time. I am trying to conn..
I wrote some code that connects to the Asterisk AMI to see if the channel is up by doing an Action status followed by a Channel: $CHANNEL. Most of the time if the channel is *NOT* up I will get:line9: Response: Error line10: Message: No such channel..
Im curently setting a lab environment for load testing an Asterisk instance.This environment includes:- a management workstation where I would like to run scripts andstore test reports- a box hosting SIPp- the Asterisk box Im load testing (System Un..
I just noticed this upon startup since updating from 15.6.1 to 16.0.0 – do any of these matter? [Oct 18 12:12:18] WARNING[4489]: loader.c:2228 load_modules: Some non-required modules failed to load. [Oct 18 12:12:18] ERROR[4489]: loader.c:2243 load_modul..
Lets say I have a conference room of 8 users. At some point in the evening, we need to hook up with a Zoom conference. That means hooking up that existing pool of users to a new PJSIP channel. An admin would dial in, enter a pin, and initiate that connecti..
It seems that app_swift does not work with Asterisk 15 or 16. I just get errors when trying to compile: [root@pbxoficina app_swift]# ./configure checking gcc… checking swift… checking asterisk… creating Makefile *****************************************************..