Hello, can you run with debug=3 and see if the function is actually executed?
Cheers, Daniel On 18/05/15 12:31, José Seabra wrote: > Hello, > > I'm using the function sdp_remove_codecs_by_id from sdpops module in > order to remove some codecs in INVITE request before send it to > freeswitch, but the function doesn't remove the codec, and it doesn't > give any error message. > > I'm using this function in request route. > > > Kamailio version is 4.2.2. > > > INVITE that kamailio receives from phone: > > INVITE sip:401@teste.d <mailto:sip%3a...@teste.itcenter.com.pt>emo.pt > <http://emo.pt>;user=phone SIP/2.0 > Record-Route: > <sip:10.0.20.102:5062;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> > Record-Route: > <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> > Via: SIP/2.0/UDP > 10.0.20.102:5062;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0 > Via: SIP/2.0/UDP > 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060 > From: "301" <sip:3...@teste.demo.pt > <mailto:sip%3a...@teste.itcenter.com.pt>>;tag=oztyflbzbx > To: <sip:4...@teste.demo.pt > <mailto:sip%3a...@teste.itcenter.com.pt>;user=phone> > Call-ID: 3c3a58a25d63-ghfc5xdg1sn0 > CSeq: 1 INVITE > Max-Forwards: 69 > Contact: > <sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1 > X-Serialnumber: 000413262FA0 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom370/8.4.35 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO, UPDATE > Allow-Events: talk, hold, refer, call-info > Supported: timer, 100rel, replaces, from-change > Call-Info: <sip:teste.demo.pt > <http://teste.itcenter.com.pt>>;appearance-index=1 > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 391 > v=0 > o=root 24935823 24935823 IN IP4 192.168.10.147 > s=call > c=IN IP4 192.168.10.147 > t=0 0 > m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101 > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:99 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:4 G723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > > > > > INVITE that kamailio send to freeswitch after execute > sdp_remove_codecs_by_id("18"): > > > INVITE sip:4...@teste.demo.pt > <mailto:sip%3a...@teste.demo.pt>;user=phone SIP/2.0. > Record-Route: > <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>. > Record-Route: > <sip:10.0.20.102:5062;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. > Record-Route: > <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. > Via: SIP/2.0/UDP > 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0. > Via: SIP/2.0/UDP > 10.0.20.102:5062;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0. > Via: SIP/2.0/UDP > 192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060. > From: "301" <sip:3...@teste.demo.pt > <mailto:sip%3a...@teste.demo.pt>>;tag=zvjgcz9zs9. > To: <sip:4...@teste.demo.pt <mailto:sip%3a...@teste.demo.pt>;user=phone>. > Call-ID: 3c3a7c84e065-pr2hm0uk9yfz. > CSeq: 2 INVITE. > Max-Forwards: 68. > Contact: > <sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1. > X-Serialnumber: 000413262FA0. > P-Key-Flags: resolution="31x13", keys="4". > User-Agent: snom370/8.4.35. <http://8.4.35.> > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, > PRACK, MESSAGE, INFO, UPDATE. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Call-Info: <sip:teste.itcenter.com.pt > <http://teste.itcenter.com.pt>>;appearance-index=1. > Session-Expires: 3600;refresher=uas. > Min-SE: 90. > Content-Type: application/sdp. > Content-Length: 403. > . > v=0. > o=root 228603317 <tel:228603317> 228603317 <tel:228603317> IN IP4 > 100.64.250.4. > s=call. > c=IN IP4 100.64.250.4. > t=0 0. > m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:8 PCMA/8000. > a=rtpmap:9 G722/8000. > a=rtpmap:99 G726-32/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:4 G723/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > a=rtcp:49405. > > > SDP body has no changes related with codecs. > > > Anyone call help please. > > Thank you > BR > José Seabra > -- > Cumprimentos > José Seabra > > > _______________________________________________ > 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 http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
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