Hi guys, I have spent quite a while reading past questions about this and I just can’t figure it out. I’m new to Asterisk.
I’m running Asterisk 13.20.0 on raspbx (pid = 13975)
I can receive calls to my Xlite softphone just fine, but it won’t connect the outbound calls. I’m hoping this is just a simple configuration change, but whatever it is, it’s got me beat… It always returns ‘Unable to create channel of type ‘SIP’ (Cause 20 - Subscriber absent)’
[2018-04-28 22:29:31] VERBOSE[28388][C-0000000d] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2018-04-28 22:29:31] VERBOSE[28388][C-0000000d] pbx.c: Executing [s@macro-dialout-trunk:25] NoOp("PJSIP/200-0000000d", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20") in new stack
Perhaps check your outbound route and Trunk outgoing settings. Did it ever work?
I was using pjsip instead of sip - I’m trying to switch but this is what I’m getting now:
[2018-04-28 23:19:15] WARNING[4925] res_pjsip/pjsip_options.c: Unable to find an endpoint to qualify contact sip:[email protected]:57446;rinstance=7ca6eff11140b922. Deleting this contact [2018-04-28 23:21:12] WARNING[5227] res_pjsip_registrar.c: Endpoint 'anonymous' has no configured AORs
Yes, you are still trying to use chan_pjsip, how have you configured the ports for chan_sip and chan_pjsip? And what port are your talking to your provider on?
When you switch the extension from chan_pjsip to chan_sip, make sure to click ‘Reset’ and ‘Apply’, and only then make changes related to the extension.
Don’t forget to restart your X-Lite so you force it to re-login.