Readers,
I am having issues with passing the max-forward header to my freeswitch service.
Here are the sip invites
Here is a call coming from a kazoo box though my carrier Into kamailio. Kamailio shows it has Max-Forward after the sanitize check, but when it reaches freeswitch no Max-Forward Header.
*KAMAILIO*
kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060 http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/ SIP/2.0 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: sip:+1NUMBERSCRUBED@67.231.9.59;isup-oli=0;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.85 sip%3A%2B18507646071@67.231.8.85> Call-ID: 537168748_70701748@67.231.9.# 537168748_70701748@67.231.9.59## CSeq: 1289511105 INVITE *Max-Forwards: 49* Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer X-FS-Support: update_display,send_info Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>;privacy=off;screen=no
v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30
*FREESWITCH* recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854: ------------------------------------------------------------------------ INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060 http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/ SIP/2.0 Record-Route: sip:10.1.13.###;lr=on;ftag=gK043cdaa8 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0 Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.# sip%3A%2B1NUMBERSCRUBED@67.231.9.59##;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.# sip%3A%2B18507646071@67.231.8.85##> Call-ID: 537168748_70701748@67.231.9.# 537168748_70701748@67.231.9.59## CSeq: 1289511105 INVITE MaxForwards would go here? Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>;privacy=off;screen=no X-AUTH-IP: 67.231.8.195
v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30
Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0
Can anyone point me in the right direction, and tell me why freeswitch isn't being passed this header? Both kamailio and freeswitch are on the same box in this example.
Thanks for your help. Mike
Hello,
most probably Max-Forward matches this:
remove_hf_re("X-.*");
Iirc, the regexp is case insensitive. You should use:
remove_hf_re("^X-.*");
In this way you are sure you don't match any "X-" inside header name.
Cheers, Daniel
On 29/10/14 16:29, Mike Dunton wrote:
Readers,
I am having issues with passing the max-forward header to my freeswitch service.
Here are the sip invites
Here is a call coming from a kazoo box though my carrier Into kamailio. Kamailio shows it has Max-Forward after the sanitize check, but when it reaches freeswitch no Max-Forward Header.
*KAMAILIO*
kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060 http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/ SIP/2.0 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.59 mailto:sip%3A%2B1NUMBERSCRUBED@67.231.9.59;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.85 mailto:sip%3A%2B18507646071@67.231.8.85> Call-ID: 537168748_70701748@67.231.9.# mailto:537168748_70701748@67.231.9.59## CSeq: 1289511105 INVITE *Max-Forwards: 49* Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer X-FS-Support: update_display,send_info Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>;privacy=off;screen=no
v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30
*FREESWITCH
recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854:
INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060 http://sip:+18507646071@aio07-bandwidth.tresta-aio.com:5060/ SIP/2.0 Record-Route: sip:10.1.13.###;lr=on;ftag=gK043cdaa8 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Record-Route: sip:67.231.8.###;lr=on;ftag=gK043cdaa8 Accept: application/sdp Allow: INVITE,ACK,CANCEL,BYE Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0 Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0 Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0 Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4 From: <sip:+1NUMBERSCRUBED@67.231.9.# mailto:sip%3A%2B1NUMBERSCRUBED@67.231.9.59##;isup-oli=0>;tag=gK043cdaa8 To: <sip:+1850764####@67.231.8.# mailto:sip%3A%2B18507646071@67.231.8.85##> Call-ID: 537168748_70701748@67.231.9.# mailto:537168748_70701748@67.231.9.59## CSeq: 1289511105 INVITE MaxForwards would go here? Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Supported: precondition Content-Length: 359 Content-Type: application/sdp Remote-Party-ID: <sip:+1NUMBERSCRUBED@67.231.9.###:5060 http://sip:+1NUMBERSCRUBED@67.231.9.59:5060/>;privacy=off;screen=no X-AUTH-IP: 67.231.8.195
v=0 o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59 s=SIP Media Capabilities c=IN IP4 67.231.9.72 t=0 0 m=audio 23918 RTP/AVP 9 0 18 96 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30
Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0
Can anyone point me in the right direction, and tell me why freeswitch isn't being passed this header? Both kamailio and freeswitch are on the same box in this example.
Thanks for your help. Mike
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Mike Dunton