I have installed FreePBX 16 with Asterisk 16 and PJSIP in a Proxmox Container running Debian 20
Here is the console log of the call:
[2022-12-09 22:12:53] NOTICE[15855]: res_pjsip_geolocation.c:194 handle_incoming_request: PJSIP/<PROVIDER>-main_account-00000023: Endpoint has no geoloc_incoming_call_profile. Done.
-- Executing [<MY_DID>@from-pstn:1] Set("PJSIP/<PROVIDER>-main_account-00000023", "__FROM_DID=<MY_DID>") in new stack
-- Executing [<MY_DID>@from-pstn:2] NoOp("PJSIP/<PROVIDER>-main_account-00000023", "Received an unknown call with DID set to <MY_DID>") in new stack
-- Executing [<MY_DID>@from-pstn:3] Goto("PJSIP/<PROVIDER>-main_account-00000023", "s,a2") in new stack
-- Goto (from-pstn,s,2)
-- Executing [s@from-pstn:2] Answer("PJSIP/<PROVIDER>-main_account-00000023", "") in new stack
> 0x7fdce00a67b0 -- Strict RTP learning after remote address set to: 208.100.60.17:17422
> 0x7fdce00a67b0 -- Strict RTP switching to RTP target address 208.100.60.17:17422 as source
[2022-12-09 22:12:53] ERROR[4080][C-00000018]: pbx_functions.c:651 ast_func_read2: Function SIP_HEADER not registered
-- Executing [s@from-pstn:3] Log("PJSIP/<PROVIDER>-main_account-00000023", "WARNING,Friendly Scanner from ") in new stack
[2022-12-09 22:12:53] WARNING[4080][C-00000018]: Ext. s:3 @ from-pstn: Friendly Scanner from
-- Executing [s@from-pstn:4] Wait("PJSIP/<PROVIDER>-main_account-00000023", "2") in new stack
-- Executing [s@from-pstn:5] Playback("PJSIP/<PROVIDER>-main_account-00000023", "ss-noservice") in new stack
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'ss-noservice.ulaw' (language 'en')
> 0x7fdce00a67b0 -- Strict RTP learning complete - Locking on source address 208.100.60.17:17422
-- Executing [s@from-pstn:6] SayAlpha("PJSIP/<PROVIDER>-main_account-00000023", "<MY_DID>") in new stack
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- <PJSIP/<PROVIDER>-main_account-00000023> Playing 'digits/<digit>.ulaw' (language 'en')
-- Executing [s@from-pstn:7] Hangup("PJSIP/<PROVIDER>-main_account-00000023", "") in new stack
== Spawn extension (from-pstn, s, 7) exited non-zero on 'PJSIP/<PROVIDER>-main_account-00000023'
-- Executing [h@from-pstn:1] Macro("PJSIP/<PROVIDER>-main_account-00000023", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/<PROVIDER>-main_account-00000023", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/<PROVIDER>-main_account-00000023", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("PJSIP/<PROVIDER>-main_account-00000023", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/<PROVIDER>-main_account-00000023' in macro 'hangupcall'
== Spawn extension (from-pstn, h, 1) exited non-zero on 'PJSIP/<PROVIDER>-main_account-00000023'
Best Answer
Replace “the_correct_DID_is_here_and_matches” with “MY_DID”
and It should have been 7582 and I entered 7852.
Why incoming and outgoing calls are not coming?
you were basically given poor advice as no one thought that your trunk and your endpoints would be under different tech. I’m going to go back to my original post, you should have just made a PJSIP trunk.
So now, go back and revert the ports the way they were (PJSIP on 5060 and Chan_SIP on 5160). Create a PJSIP Trunk, under the PJSIP Settings
Authentication: None
SIP Server URI: 67.231.8.195
SIP Server Port: 5060
Advanced tab where it says Match (Permit) put : 65.213.8.195/32,67.231.4.195/32
This trunk will send calls to the 67.231.8.195 server and accept calls from both.