As a very new user to FreePBX, but through tutorials, lots of reading and some trial and error, i managed to configure extensions, make internal calls and even configure an external VOIP provider and make external calls using that.
However, when configuring a SIP Trunk to our local provider, i seem to have hit a wall…
Evertthing seems to be well configured, but i get the following message when i restart the services (fwconsole restart) and enter verbose mode (asterisk -rvvv):
Contact Altice/sip:XXX.XXX.XXX.XXX:5060 is now Unreachable. RTT: 0.000 msec
We have an Alcatel OXE which we are trying to replace, and it does work, so network problems don’t seem to be the case.
I’m basically out of ideas.
Maybe someone has a fresh perspective?
After further digging, i’ve found the following responses:
[2020-11-18 11:06:31] WARNING: res_pjsip_outbound_registration.c:829 schedule_retry: No response received from ‘sip:XXX.XXX.XXX.XXX:5060’ on registration attempt to ‘sip:Altice@XXX.XXX.XXX.XXX:5060’, retrying in ‘60’
And if i try to place a call, i get a busy message, and also this output:
[2020-11-18 11:10:46] ERROR: res_pjsip.c:3562 ast_sip_create_dialog_uac: Endpoint ‘Altice’: Could not create dialog to invalid URI ‘Altice’. Is endpoint registered and reachable?
[2020-11-18 11:10:46] ERROR: chan_pjsip.c:2687 request: Failed to create outgoing session to endpoint ‘Altice’
[2020-11-18 11:10:46] WARNING[C-00000001]: app_dial.c:2576 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
The server is definitely reachable.
So, the prblem is either:
-Bad config (i used PJSIP, but i should try with SIP?..)
-Network problem (incoming)