Inbound Calls getting SIP error 404

radi27 · December 16, 2021, 3:54am

Hello guys,

Not sure if the question has been handled before - if yes, please excuse me.

I have setup Asterisk 13.35.0 on a VPS with MagnusBilling installed.
I also have a Callcentric account with 2 SIP extensions. First one - default is configured as Trunk in Magnusbilling to be used for incoming calls, as per the guide here: Asterisk 14 Configuration and Review . If I don’t add the extra hosts(outlined in the guide, despite it being for version 14), Asterisk responds with 401 to SIP invites.

However, I am getting a 404 error when dialing the Callcentric DID from the other extension. It is registered, and I am getting the SIP invite.

This is the log of a test call with verbose and debug set to 99 from the asterisk log file. In console, nothing happens.

[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: = Looking for  Call ID: E584BBDA5B443121C1EC6F1548AB072160E07AC2 (Checking From) --From tag D8C868E1B5840A4E9B12A71E475F654A --To-tag
[2021-12-16 06:38:59] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.38', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:38:59] DEBUG[2342] netsock2.c: Splitting 'CALLCENTRIC.OWN.SUBNET.38:5060' into...
[2021-12-16 06:38:59] DEBUG[2342] netsock2.c: ...host 'CALLCENTRIC.OWN.SUBNET.38' and port '5060'.
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for E584BBDA5B443121C1EC6F1548AB072160E07AC2 - INVITE (No RTP)
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, path"
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] sip/reqresp_parser.c: Found SIP option: -replaces-
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] sip/reqresp_parser.c: Matched SIP option: replaces
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] sip/reqresp_parser.c: Found SIP option: -path-
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] sip/reqresp_parser.c: Matched SIP option: path
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: Splitting 'CALLCENTRIC.OWN.SUBNET.38:5060' into...
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: ...host 'CALLCENTRIC.OWN.SUBNET.38' and port '5060'.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: Splitting 'callcentric.com' into...
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: ...host 'callcentric.com' and port ''.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f3948017e88'
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] res_rtp_asterisk.c: Allocated port 11984 for RTP instance '0x7f3948017e88'
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: RTP instance '0x7f3948017e88' is setup and ready to go
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: Splitting 'my.vps.fqdn.com' into...
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: ...host 'my.vps.fqdn.com' and port ''.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f3948017e88'
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Setting NAT on RTP to On
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing session-level SDP o=- 1274635394 5418 IN IP4 CALLCENTRIC.OWN.SUBNET.38... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing session-level SDP s=sawcrip... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: Splitting 'CALLCENTRIC.OWN.SUBNET.38' into...
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: ...host 'CALLCENTRIC.OWN.SUBNET.38' and port ''.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing session-level SDP c=IN IP4 CALLCENTRIC.OWN.SUBNET.38... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 9 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 103 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 102 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 18 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 3 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Setting tx payload type 101 based on m type on 0x7f38f543f140
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 ILBC/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:103 opus/48000/2... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:102 mode=20... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:103 maxplaybackrate=16000;maxaveragebitrate=24000;useinbandfec=1;usedtx=1... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=zrtp-hash:1.10 48D15F83397B493C5808D28D9BC89B4E1ABCFCE8E1A1711748430BF606671786... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=silenceSupp:off - - - -... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Processing media-level (audio) SDP a=setup:actpass... UNSUPPORTED OR FAILED.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.38', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f3948017e88'
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 0 (0x7f3948001220) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 3 (0x7f394800d1e0) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 8 (0x7f394800d210) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 9 (0x7f394800dea0) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 18 (0x7f394800ded0) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 101 (0x7f394800f960) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 102 (0x7f39480057f0) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] rtp_engine.c: Copying payload 103 (0x7f394800d620) from 0x7f38f543f140 to 0x7f3948018050
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f3948017e88'
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: We're settling with these formats: (ulaw)
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Checking SIP call limits for device MY-CALLCENTRIC-DID-NUMBER
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Updating call counter for incoming call
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: Splitting 'MY.VPS.OWN.IP:5060' into...
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: ...host 'MY.VPS.OWN.IP' and port ''.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: Splitting 'callcentric.com' into...
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] netsock2.c: ...host 'callcentric.com' and port ''.
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.38:5060
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Updating call counter for incoming call
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Auto destroying SIP dialog '1388852203-790239605-983204601'
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Destroying SIP dialog 1388852203-790239605-983204601
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 244 ms (t1 122 ms (Retrans id #59))
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.38:5060
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: = Looking for  Call ID: E584BBDA5B443121C1EC6F1548AB072160E07AC2 (Checking From) --From tag D8C868E1B5840A4E9B12A71E475F654A --To-tag as53e9c3da
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2021-12-16 06:38:59] DEBUG[2342][C-00000003] chan_sip.c: Stopping retransmission on 'E584BBDA5B443121C1EC6F1548AB072160E07AC2' of Response 2: Match Found
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Destroying SIP dialog E584BBDA5B443121C1EC6F1548AB072160E07AC2
[2021-12-16 06:38:59] DEBUG[2342] rtp_engine.c: Destroyed RTP instance '0x7f3948017e88'
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for 0a3bc0b07f6ead213c26da5e066db272@MY.VPS.OWN.IP:5060 - OPTIONS (No RTP)
[2021-12-16 06:38:59] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.134', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: SIP call-id changed from '0a3bc0b07f6ead213c26da5e066db272@MY.VPS.OWN.IP:5060' to '3b9fecb06df6075c4a41aad02632f189@MY.VPS.OWN.IP:5060'
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Initializing initreq for method OPTIONS - callid 3b9fecb06df6075c4a41aad02632f189@MY.VPS.OWN.IP:5060
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.134:5060
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for 2394415342f809c80fe826dc674c76c4@MY.VPS.OWN.IP:5060 - OPTIONS (No RTP)
[2021-12-16 06:38:59] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.133', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: SIP call-id changed from '2394415342f809c80fe826dc674c76c4@MY.VPS.OWN.IP:5060' to '537094d06e2bce7636da075e78a8f4d8@MY.VPS.OWN.IP:5060'
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Initializing initreq for method OPTIONS - callid 537094d06e2bce7636da075e78a8f4d8@MY.VPS.OWN.IP:5060
[2021-12-16 06:38:59] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.133:5060
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for 4a9360193523a30804d498fa5468ef6f@MY.VPS.OWN.IP:5060 - OPTIONS (No RTP)
[2021-12-16 06:39:00] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.132', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: SIP call-id changed from '4a9360193523a30804d498fa5468ef6f@MY.VPS.OWN.IP:5060' to '668b8188343242ef1b09044217d71e59@MY.VPS.OWN.IP:5060'
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: Initializing initreq for method OPTIONS - callid 668b8188343242ef1b09044217d71e59@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.132:5060
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.134:5060
[2021-12-16 06:39:00] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.133:5060
[2021-12-16 06:39:01] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.132:5060
[2021-12-16 06:39:01] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.134:5060
[2021-12-16 06:39:01] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.133:5060
[2021-12-16 06:39:02] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.132:5060
[2021-12-16 06:39:02] DEBUG[2547] manager.c: Running action 'Login'
[2021-12-16 06:39:02] DEBUG[2547] manager.c: Running action 'Command'
[2021-12-16 06:39:02] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.134:5060
[2021-12-16 06:39:02] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.133:5060
[2021-12-16 06:39:03] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.132:5060
[2021-12-16 06:39:03] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.134:5060
[2021-12-16 06:39:03] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.133:5060
[2021-12-16 06:39:03] DEBUG[2342] chan_sip.c: Destroying SIP dialog 3b9fecb06df6075c4a41aad02632f189@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:03] DEBUG[2342] chan_sip.c: Destroying SIP dialog 537094d06e2bce7636da075e78a8f4d8@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:04] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.132:5060
[2021-12-16 06:39:04] DEBUG[2342] chan_sip.c: Destroying SIP dialog 668b8188343242ef1b09044217d71e59@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:06] DEBUG[2549] manager.c: Running action 'Login'
[2021-12-16 06:39:06] DEBUG[2549] manager.c: Running action 'Command'
[2021-12-16 06:39:10] DEBUG[2552] manager.c: Running action 'Login'
[2021-12-16 06:39:10] DEBUG[2552] manager.c: Running action 'Command'
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for 5fc9476d1510b5406d2dc47a436d02bd@MY.VPS.OWN.IP:5060 - OPTIONS (No RTP)
[2021-12-16 06:39:13] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.134', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: SIP call-id changed from '5fc9476d1510b5406d2dc47a436d02bd@MY.VPS.OWN.IP:5060' to '438617505ae2f37333696b242541a599@MY.VPS.OWN.IP:5060'
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Initializing initreq for method OPTIONS - callid 438617505ae2f37333696b242541a599@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.134:5060
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for 6f210d9b2cb5360d1b5b061e4170452b@MY.VPS.OWN.IP:5060 - OPTIONS (No RTP)
[2021-12-16 06:39:13] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.133', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: SIP call-id changed from '6f210d9b2cb5360d1b5b061e4170452b@MY.VPS.OWN.IP:5060' to '1365c309736660b534e29bda5fedb87b@MY.VPS.OWN.IP:5060'
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Initializing initreq for method OPTIONS - callid 1365c309736660b534e29bda5fedb87b@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:13] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.133:5060
[2021-12-16 06:39:14] DEBUG[2342] chan_sip.c: Allocating new SIP dialog for 3061f93d7b8fdd075a0f89176d02f92e@MY.VPS.OWN.IP:5060 - OPTIONS (No RTP)
[2021-12-16 06:39:14] DEBUG[2342] acl.c: For destination 'CALLCENTRIC.OWN.SUBNET.132', our source address is 'MY.VPS.OWN.IP'.
[2021-12-16 06:39:14] DEBUG[2342] chan_sip.c: Setting AST_TRANSPORT_UDP with address MY.VPS.OWN.IP:5060
[2021-12-16 06:39:14] DEBUG[2342] chan_sip.c: SIP call-id changed from '3061f93d7b8fdd075a0f89176d02f92e@MY.VPS.OWN.IP:5060' to '6733bb28361f64a664046be06d6a2836@MY.VPS.OWN.IP:5060'
[2021-12-16 06:39:14] DEBUG[2342] chan_sip.c: Initializing initreq for method OPTIONS - callid 6733bb28361f64a664046be06d6a2836@MY.VPS.OWN.IP:5060
[2021-12-16 06:39:14] DEBUG[2342] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for CALLCENTRIC.OWN.SUBNET.132:5060

I have replaced identifying data with placeholders. The DID is routed to IVR on my system, but no matter what I set it to - same error.

Please let me know if you require more information on debugging this issue. It is driving me crazy. I tried other provider as well (again SIP account as trunk), but it’s giving me same error. Also that one’s a bit too complex to setup, because it requires SIP proxy(which I know how to setup on an IP phone, but not when configuring the account as SIP trunk). P.S. Perhaps that requires “outbound_proxy” directive?

Thanks in advance for any help.

david551 · December 16, 2021, 10:34am

Asterisk 13 is about two months beyond full end of life. Please try with the latest sub-version of a supported version.

chan_sip is deprecated and only receives community support, and then only in theory. Please try with chan_pjsip.

Please provide the full log, not the debug log. There is likely to be a line saying extension not found in context. Generally the full log, at verbosity 5, and the “pjsip set logger on” output will be sufficient for this sort of problem, along with pjsip.conf and any files included from it. (Make suitable substitutions for deprecated drivers.

radi27 · December 16, 2021, 10:49am

Unfortunately, I cannot use a new version as the billing solution relies on asterisk 13 - it installs it by default and supports chan_sip. I looked into pjsip as an alternative, but I am not sure how my billing solution can provision extensions in that case. So, I am left with chan_sip only.

david551 · December 16, 2021, 11:15am

Contact the billing solution provider. However, I suspect it won’t change the “not found in context” message you will get from the full log, and which will indicate whether the problem is with the request URI user, the context of the peer, or the dialplan for that context.

radi27 · December 16, 2021, 11:46am

I can’t afford their support. Is there any way to get more logging for chan_sip to track the reason for rejected calls?

radi27 · December 19, 2021, 9:08pm

I managed to get it work with the Asterisk 14 guide of Callcentric after a reinstall of the whole system. For those who come by this thread:

You need to add the “host” variables of all their machines added and do not set outbound_proxy(I made the mistake to try to set it, as per the instructions for a SIP phone). Make sure also that fromъser and fromdomain are set as well.

The register string is the default and type should be left peer, despite the guide saying “friend” for incoming calls.

Not sure why it went wrong on the first try. Thread can be closed.