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
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>;tag=oztyflbzbx
To: <sip:401@teste.demo.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>;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


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