Hello,

I added the following xlog before function

xlog("L_INFO", "[SDPOPS] executing function sdp_remove_codecs_by_id($avp(s:codecs_to_remove)) ID=$ci\n");

and if i have set debug=2 I can see the xlog message, if I change it to 3 I cannot see my message, even if I set xlog with L_DBG i cannot see the message in syslog, this is a weird behavior, can be something wrong with my rsyslog service?

I did a test that was edit the c function sdp_remove_codecs_by_id in sdpops_mod.c and i changed the log message from LM_DBG to LM_INFO, then i compiled and ran again the test and i can see the internal log messages from function sdp_remove_codecs_by_id.

syslog:

May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Call flow id '18' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Call flow name 'queue test' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Action Type 'CallQueue' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Has object left '0' ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Call Queues - R=sip:400@test.centrex.coditel.be;user=phone  ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: Relaying request to freeSWITCH, M=INVITE, du='sip:10.0.20.26:5060',F=sip:201@test.centrex.coditel.be - R=sip:400@test.centrex.coditel.be;user=phone ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: <script>: [SDPOPS] executing function sdp_remove_codecs_by_id(18) ID=3134333230343038333536373236-i0wa7mreng1w
May 19 15:07:17 ccp2 /usr/local/sbin/kamailio[9068]: INFO: sdpops [sdpops_mod.c:331]: sdp_remove_codecs_by_id(): attempting to remove codecs from sdp: [18]



BR
José Seabra

2015-05-19 12:25 GMT+01:00 Daniel-Constantin Mierla <miconda@gmail.com>:
Hello,

can you enable cfgtrace via debugger module or add an xlog just before calling the function in configuration file and see if related message appears in syslog?

Cheers,
Daniel


On 19/05/15 11:05, José Seabra wrote:
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.demo.pt;user=phone SIP/2.0
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>;tag=oztyflbzbx
To: <sip:401@teste.demo.pt;user=phone>
Call-ID: 3c3a58a25d63-ghfc5xdg1sn0
CSeq: 1 INVITE
Max-Forwards: 69
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>;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.
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.
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


_______________________________________________
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sr-users@lists.sip-router.org
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-- 
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

_______________________________________________
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




--
Cumprimentos
José Seabra

-- 
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



--
Cumprimentos
José Seabra