Hello, whenever i make OB Calls I get All circuits are busy response.
This is my logs
[2022-11-25 00:58:38] ERROR[2134] res_pjsip.c: Endpoint ‘Trunk_OB’: Could not create dialog to invalid URI ‘Trunk_OB’. Is endpoint registered and reachable?
7513 [2022-11-25 00:58:38] ERROR[2134] chan_pjsip.c: Failed to create outgoing session to endpoint ‘Trunk_OB’
7514 [2022-11-25 00:58:38] WARNING[24678][C-0000000b] app_dial.c: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
7515 [2022-11-25 00:58:38] VERBOSE[24678][C-0000000b] app_dial.c: No devices or endpoints to dial (technology/resource)
7516 [2022-11-25 00:58:38] VERBOSE[24678][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:37] NoOp(“PJSIP/8077-0000000b”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3”) in new stack
Does the Asterisk log show any errors related to registration or unreachable?
Does Reports → Asterisk Info → Registries show the trunk as registered?
Are incoming calls working properly?
Hi Stewart, I’ve made progress in the issue I just fixed the sip server IP, now it’s giving me number is not in service.
about the incoming calls, I haven’t made the setup yet currently I just need to make OB at this moment
At the Asterisk command prompt, type pjsip set logger on
or if for some reason you are using chan_sip trunks, type sip set debug on
attempt an outbound call, paste the Asterisk log for the call at pastebin.freepbx.org and post the link here. If you are too new to post links, just post the last eight hex characters of the URL.
Only the first execif condition was satisfied, so caller ID was set to 21651263350. Was that (apparently Orange mobile) number assigned by your provider?
Please paste the complete log for the call, including pjsip logger, as previously requested.
On line 170, the call is sent to the trunk, but the next line shows an incoming call to the same number. Assuming that you weren’t calling yourself, this is likely a trunk misconfiguration.
Unfortunately, pjsip logger was off so we can’t see the SIP traffic. Please enable it and paste another log.