Hello, Thank you for your reply
I ran kamailio with debug=3 and log_stderror=yes and the only thing that i see related with function sdp_remove_codecs_by_id is:
0(4707) DEBUG: <core> [route.c:907]: fix_actions(): fixing sdp_remove_codecs_by_id()
if i set debug=3 and log_stderror=no then i look for syslog file where kamailio is writting logs, and i don't see anything related with function sdp_remove_codecs_by_id.
I'm not using msg_apply_changes function.
Thank you for your support
BR José Seabra
2015-05-18 13:26 GMT+01:00 Daniel-Constantin Mierla miconda@gmail.com:
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 sip%3A401@teste.itcenter.com.ptemo.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:301@teste.demo.pt sip%3A301@teste.itcenter.com.pt
;tag=oztyflbzbx
To: <sip:401@teste.demo.pt sip%3A401@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:401@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:301@teste.demo.pt;tag=zvjgcz9zs9. To: sip:401@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. 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;appearance-index=1. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 403. . v=0. o=root 228603317 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 listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
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