David, is this the correct invite info from asterisk to bandwidth
On this example the caller ID is set on the extension
“myname” <9165551212
Line 7734 it shows From: “PBX1 101” (PBX1 is the server and 101 is the extension
Line 7767 show the same as 7734
Line 7938 shows From: “myname” sip:[email protected]…
Line 7947 shows P-Asserted-Identity: “myname” <sip:9165551212@…
Line 7943, 7983 show the same as 7938
7731 [2022-05-15 02:05:00] VERBOSE[2737] res_pjsip_logger.c: <— Received SIP request (1151 bytes) from UDP:10.0.0.18:5060 —>
7732 INVITE sip:[email protected]:5160 SIP/2.0
7733 Via: SIP/2.0/UDP 10.0.0.18:5060;branch=z9hG4bK2681100091
7734 From: “PBX1 101” sip:[email protected]:5160;tag=496803802
7735 To: sip:[email protected]:5160
7736 Call-ID: [email protected]
7737 CSeq: 2 INVITE
7738 Contact: sip:[email protected]:5060
7740 Content-Type: application/sdp
7741 Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
7742 Max-Forwards: 70
7743 User-Agent: Yealink SIP-T27P 45.82.0.30
7744 Allow-Events: talk,hold,conference,refer,check-sync
7745 Supported: replaces
7746 Content-Length: 286
7747
7748 v=0
7749 o=- 20158 20158 IN IP4 10.0.0.18
7750 s=SDP data
7751 c=IN IP4 10.0.0.18
7752 t=0 0
7753 m=audio 12428 RTP/AVP 0 8 18 9 101
7754 a=rtpmap:0 PCMU/8000
7755 a=rtpmap:8 PCMA/8000
7756 a=rtpmap:18 G729/8000
7757 a=fmtp:18 mode=20
7758 a=rtpmap:9 G722/8000
7759 a=sendrecv
7760 a=rtpmap:101 telephone-event/8000
7761 a=fmtp:101 0-15
7762
7763 [2022-05-15 02:05:00] VERBOSE[2738] res_pjsip_logger.c: <— Transmitting SIP response (296 bytes) to UDP:10.0.0.18:5060 —>
7764 SIP/2.0 100 Trying
7765 Via: SIP/2.0/UDP 10.0.0.18:5060;rport=5060;received=10.0.0.18;branch=z9hG4bK2681100091
7766 Call-ID: [email protected]
7767 From: “PBX1 101” sip:[email protected];tag=496803802
7768 To: sip:[email protected]
7769 CSeq: 2 INVITE
7770 Server: FPBX-15.0.23(16.17.0)
7771 Content-Length: 0
7772
7773
7936 INVITE sip:[email protected] SIP/2.0
7937 Via: SIP/2.0/UDP 73.220.172.215:5160;rport;branch=z9hG4bKPj5a6c2646-a92a-4ac0-8315-906d8d5c326e
7938 From: “myname” sip:[email protected];tag=241ef512-5e4e-4a9b-ac32-3eb2f11826e5
7939 To: sip:[email protected]
7940 Contact: sip:[email protected]:5160
7941 Call-ID: a7179b9a-536d-43e8-8739-09edb2e444f3
7942 CSeq: 11919 INVITE
7943 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
7944 Supported: 100rel, timer, replaces, norefersub, histinfo
7945 Session-Expires: 1800
7946 Min-SE: 90
7947 P-Asserted-Identity: “myname” sip:[email protected]
7948 Max-Forwards: 70
7949 User-Agent: FPBX-15.0.23(16.17.0)
7950 Content-Type: application/sdp
7951 Content-Length: 341
7952
7953 v=0
7954 o=- 577021793 577021793 IN IP4 73.220.172.215
7955 s=Asterisk
7956 c=IN IP4 73.220.172.215
7957 t=0 0
7958 m=audio 12928 RTP/AVP 0 8 9 3 111 101
7959 a=rtpmap:0 PCMU/8000
7960 a=rtpmap:8 PCMA/8000
7961 a=rtpmap:9 G722/8000
7962 a=rtpmap:3 GSM/8000
7963 a=rtpmap:111 G726-32/8000
7964 a=rtpmap:101 telephone-event/8000
7965 a=fmtp:101 0-16
7966 a=ptime:20
7967 a=maxptime:150
7968 a=sendrecv
7969
7970 [2022-05-15 02:05:01] VERBOSE[2737] res_pjsip_logger.c: <— Received SIP response (341 bytes) from UDP:67.123.123.123:5060 —>
7971 SIP/2.0 100 Trying
7972 Via: SIP/2.0/UDP 10.0.0.101:5160;branch=z9hG4bKPj5a6c2646-a92a-4ac0-8315-906d8d5c326e;rport=5160
7973 From: “myname” sip:[email protected];tag=241ef512-5e4e-4a9b-ac32-3eb2f11826e5
7974 To: sip:[email protected];tag=gK0ca47c4e
7975 Call-ID: a7179b9a-536d-43e8-8739-09edb2e444f3
7976 CSeq: 11919 INVITE
7977 Content-Length: 0
7978
7979
7980 [2022-05-15 02:05:02] VERBOSE[2737] res_pjsip_logger.c: <— Received SIP response (789 bytes) from UDP:67.123.123.123:5060 —>
7981 SIP/2.0 183 Session Progress
7982 Via: SIP/2.0/UDP 10.0.0.101:5160;branch=z9hG4bKPj5a6c2646-a92a-4ac0-8315-906d8d5c326e;rport=5160
7983 From: “myname” sip:[email protected];tag=241ef512-5e4e-4a9b-ac32-3eb2f11826e5
7984 To: sip:[email protected];tag=gK0ca47c4e
7985 Call-ID: a7179b9a-536d-43e8-8739-09edb2e444f3
7986 CSeq: 11919 INVITE
7987 Contact: sip:[email protected]:5060
7988 Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
7989 P-Asserted-Identity: tel:+19161234567
7990 Content-Length: 230
7991 Content-Disposition: session; handling=required
7992 Content-Type: application/sdp
7993
7994 v=0
7995 o=Sonus_UAC 873683 24523 IN IP4 67.231.4.7
7996 s=SIP Media Capabilities
7997 c=IN IP4 67.231.4.7
7998 t=0 0
7999 m=audio 43944 RTP/AVP 0 101
8000 a=rtpmap:0 PCMU/8000
8001 a=rtpmap:101 telephone-event/8000
8002 a=fmtp:101 0-15
8003 a=sendrecv
8004 a=ptime:20
I’m still not getting outbound CallerID. Does this log match with what Bandwidth is saying that "they saw that the From header had “myname” on it, but the Contact header only had the originating number.
I appreciate lettings me know that I was looking at the wrong header info. I hope I got the right info this time.