18 May
2015
18 May
'15
12:31 p.m.
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(a)teste.itcenter.com.pt>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:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt>
;tag=oztyflbzbx
To: <sip:401@teste.demo.pt
<sip%3A401(a)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
18 May
18 May
1:26 p.m.
Hi
Did you use msg_apply_changes() before relaying the INVITE
?http://kamailio.org/docs/modules/4.1.x/modules/textopsx.html#textopsx.f.msg_apply_changes
Regards,Dragos
From: José Seabra <joseseabra4(a)gmail.com>
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org>
Sent: Monday, May 18, 2015 12:31 PM
Subject: [SR-Users] Function sdp_remove_codecs_by_id seems to be not working
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/2.0Record-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.0Via: SIP/2.0/UDP
192.168.10.147:5060;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060From:
"301" <sip:301@teste.demo.pt>;tag=oztyflbzbxTo:
<sip:401@teste.demo.pt;user=phone>Call-ID: 3c3a58a25d63-ghfc5xdg1sn0CSeq: 1
INVITEMax-Forwards: 69Contact:
<sip:301@192.168.10.147:5060;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1X-Serialnumber:
000413262FA0P-Key-Flags: resolution="31x13", keys="4"User-Agent:
snom370/8.4.35Accept: application/sdpAllow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS,
NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATEAllow-Events: talk, hold, refer,
call-infoSupported: timer, 100rel, replaces, from-changeCall-Info:
<sip:teste.demo.pt>;appearance-index=1Session-Expires: 3600;refresher=uasMin-SE:
90Content-Type: application/sdpContent-Length: 391v=0o=root 24935823 24935823 IN IP4
192.168.10.147s=callc=IN IP4 192.168.10.147t=0 0m=audio 19410 RTP/AVP 0 8 9 99 3 18 4
101a=rtpmap:0 PCMU/8000.a=rtpmap:8 PCMA/8000a=rtpmap:9 G722/8000a=rtpmap:99
G726-32/8000a=rtpmap:3 GSM/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:4
G723/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=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 youBRJosé Seabra
--
CumprimentosJosé Seabra
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
2:26 p.m.
New subject: Function sdp_remove_codecs_by_id seems to be not working
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%3A401@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:301@teste.demo.pt
<mailto:sip%3A301@teste.itcenter.com.pt>>;tag=oztyflbzbx
To: <sip:401@teste.demo.pt
<mailto: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
<mailto:sip%3A401@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
<mailto:sip%3A301@teste.demo.pt>>;tag=zvjgcz9zs9.
To: <sip:401@teste.demo.pt <mailto:sip%3A401@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(a)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
19 May
19 May
11:05 a.m.
New subject: Function sdp_remove_codecs_by_id seems to be not working
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(a)gmail.com>om>:
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(a)teste.itcenter.com.pt>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:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt>
--
Cumprimentos
José Seabra
;tag=oztyflbzbx
To:
<sip:401@teste.demo.pt <sip%3A401(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
1:25 p.m.
New subject: Function sdp_remove_codecs_by_id seems to be not working
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(a)gmail.com
<mailto: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:
--
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
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%3A401@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:301@teste.demo.pt
<mailto:sip%3A301@teste.itcenter.com.pt>>;tag=oztyflbzbx
To: <sip:401@teste.demo.pt
<mailto: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
<mailto:sip%3A401@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
<mailto:sip%3A301@teste.demo.pt>>;tag=zvjgcz9zs9.
To: <sip:401@teste.demo.pt
<mailto:sip%3A401@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(a)lists.sip-router.org <mailto: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://twitter.com/#%21/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(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Cumprimentos
José Seabra
3:13 p.m.
New subject: Function sdp_remove_codecs_by_id seems to be not working
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(a)gmail.com>om>:
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(a)gmail.com>om>:
--
Cumprimentos
José Seabra
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(a)teste.itcenter.com.pt>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:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt>
--
Cumprimentos
José Seabra
--
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
;tag=oztyflbzbx
To:
<sip:401@teste.demo.pt <sip%3A401(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
9 Jul
9 Jul
11:21 a.m.
New subject: Function sdp_remove_codecs_by_id seems to be not working
Hello Daniel,
Returning to this issue(I didn't have enough time before, to continue with
debug), I have configured the module debugger, in order to try see more
information when kamailio executes the function "sdp remove codecs_by_id",
and what i see is that the function is executed and the logs says that
the function removes the codecs specified in SDP body, but in reality it
doesn't happen, and there isn't any errors executing the function.
Please have a look at the logs below:
9(9082) DEBUG: <core> [db_res.c:92]: db_free_columns(): freeing result
names at 0x7fe2174741f8
9(9082) DEBUG: <core> [db_res.c:97]: db_free_columns(): freeing result
types at 0x7fe2174743c0
9(9082) DEBUG: <core> [db_res.c:52]: db_free_rows(): freeing 1 rows
9(9082) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at
0x7fe21765a298
9(9082) DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at
0x7fe21765a250
9(9082) DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set
at 0x7fe217480910
9(9082) DEBUG: registrar [lookup.c:196]: lookup_helper(): contact for [
302(a)teste.demo.pt] found by address
9(9082) DEBUG: <core> [parser/parse_rr.c:452]: get_path_dst_uri(): path
for branch:
'<sip:lb@10.0.20.103:5062;lr;received=sip:100.64.250.254:46572>'
9(9082) INFO: <script>: Internal call's for account teste.demo.pt -
R=sip:302@192.168.10.197:5060;line=y8nryy19
ID=313433363433323332313432323132-2dqr4288p1lg
9(9082) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg
id=1 , global msg id=1 , T on entrance=(nil)
9(9082) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request:
start searching: hash=51835, isACK=0
9(9082) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261
transaction matching failed
9(9082) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG:
t_lookup_request: no transaction found
9(9082) DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG:
trans=0x7fe1fd3e5c68, callback type 1, id 0 entered
9(9082) DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG:
trans=0x7fe1fd3e5c68, callback type 1, id 0 entered
9(9082) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg
0x7fe1fd3d6ed8 with 1 -> 2
9(9082) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2457 found
on entry 2631
9(9082) DEBUG: dialog [dlg_handlers.c:721]: dlg_onreq(): dialog added to
tm callbacks
9(9082) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg
0x7fe1fd3d6ed8 with 1 -> 1
9(9082) DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG:
trans=0x7fe1fd3e5c68, callback type 1, id 0 entered
9(9082) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated:
a95682a5a071560d00ceb5cf18f5bbb1
9(9082) DEBUG: <core> [forward.c:702]: update_sock_struct_from_via():
trying SRV lookup
9082(9082) DEBUG: debugger [debugger_api.c:266]: dbg_msgid_filter():
process_no:9082 indx:9
9082(9082) DEBUG: debugger [debugger_api.c:280]: dbg_msgid_filter():
msg->id:1 msgid_base:0 -> 1
9(9082) DEBUG: <script>: new branch [0] to sip:302@192.168.10.197:5060
;line=y8nryy19
*9(9082) INFO: <script>: [Callflow]
------------------------------------------codec remove
3,0,9-------------------------------------------
R=sip:302@192.168.10.197:5060;line=y8nryy19
ID=313433363433323332313432323132-2dqr4288p1lg*
* 9(9082) DEBUG: textops [textops.c:2473]: has_body_f(): content type is
196611*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:339]: sdp_remove_codecs_by_id():
attempting to remove codecs from sdp: [3,0,9]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:354]: sdp_remove_codecs_by_id():
stream 0 of 0 - payloads [9 0 8 3 99 112 18 101]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:365]: sdp_remove_codecs_by_id():
codecs [9 0 8 3 99 112 18 101] - remove [3]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:289]: sdp_remove_str_codec_id():
found codec [3] inside [9 0 8 3 99 112 18 101]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 101/telephone-event*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 18/G729*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 112/AAL2-G726-32*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 99/G726-32*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 3/GSM*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:197]:
sdp_remove_str_codec_id_attrs(): removing line: a=rtpmap:3 GSM/8000*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 8/PCMA*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 0/PCMU*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 9/G722*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:365]: sdp_remove_codecs_by_id():
codecs [9 0 8 3 99 112 18 101] - remove [0]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:289]: sdp_remove_str_codec_id():
found codec [0] inside [9 0 8 3 99 112 18 101]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 101/telephone-event*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 18/G729*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 112/AAL2-G726-32*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 99/G726-32*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 3/GSM*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 8/PCMA*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 0/PCMU*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:197]:
sdp_remove_str_codec_id_attrs(): removing line: a=rtpmap:0 PCMU/8000*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 9/G722*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:365]: sdp_remove_codecs_by_id():
codecs [9 0 8 3 99 112 18 101] - remove [9]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:289]: sdp_remove_str_codec_id():
found codec [9] inside [9 0 8 3 99 112 18 101]*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 101/telephone-event*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 18/G729*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 112/AAL2-G726-32*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 99/G726-32*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 3/GSM*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 8/PCMA*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 0/PCMU*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:190]:
sdp_remove_str_codec_id_attrs(): a= ... for codec 9/G722*
* 9(9082) DEBUG: sdpops [sdpops_mod.c:197]:
sdp_remove_str_codec_id_attrs(): removing line: a=rtpmap:9 G722/8000*
9(9082) DEBUG: <script>: start of route sip session- 1 INVITE
sip:3022@teste.demo.pt;user=phone SIP/2.0
Record-Route: <sip:10.0.20.103:5062
;r2=on;lr=on;ftag=etq6ixm8l0;nat=yes;lb=yes>
Record-Route: <sip:100.64.250.5;r2=on;lr=on;ftag=etq6ixm8l0;nat=yes;lb=yes>
Via: SIP/2.0/UDP 10.0.20.103:5062
;branch=z9hG4bKb7ac.4d1b542c3b99d7410acbc985476e83d2.0
Via: SIP/2.0/UDP 192.168.30.107:5060
;received=100.64.250.254;branch=z9hG4bK-k9wsyxdyb3sl;rport=5060
From: "301" <sip:301@teste.demo.pt>;tag=etq6ixm8l0
To: <sip:3022@teste.demo.pt;user=phone>
Call-ID: 313433363433323332313432323132-2dqr4288p1lg
CSeq: 2 INVITE
Max-Forwards: 68
User-Agent: snom370/8.7.5.17
Contact: <sip:301@192.168.30.107:5060
;alias=100.64.250.254~5060~1;line=lb5k2cyy>;reg-id=1
X-Serialnumber: 000413262FA1
P-Key-Flags: resolution="31x13", keys="4"
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
Session-Expires: 3600
Min-SE: 90
Proxy-Authorization: Digest username="301",realm="teste.demo.pt
",nonce="VZ45hFWeOFh4j60H/FTV12CfMIHUwH3/",uri="sip:3022@teste.demo.pt
;user=phone",response="c1d749e844a81cc68841c1d43be87885",algorithm=MD5
Content-Type: application/sdp
Content-Length: 407
P-Voicis-LB: 10.0.20.103
P-Voicis-LBPort: 5062
P-VOICIS-Src-Ip: 100.64.250.254
P-VOICIS-Src-Port: 5060
P-VOICIS-Src-Proto: udp
P-Auth-Info: 1
P-VOICIS-Src-Nat: 1
v=0
o=root 1097951374 1097951374 IN IP4 192.168.30.107
s=call
c=IN IP4 192.168.30.107
t=0 0
m=audio 19818 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
9(9082) DEBUG: pv [pv_core.c:376]: pv_get_xto_attr(): no Tag parameter
9(9082) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg
0x7fe1fd3d6ed8 with 1 -> 2
9(9082) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2457 found
on entry 2631
9(9082) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg
0x7fe1fd3d6ed8 with 1 -> 1
9(9082) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg
0x7fe1fd3d6ed8 with 1 -> 2
9(9082) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2457 found
on entry 2631
9(9082) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg
0x7fe1fd3d6ed8 with 1 -> 1
9(9082) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT
0x7fe216f4e380
9(9082) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg
0x7fe1fd3d6ed8 with 1 -> 2
9(9082) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2457 found
on entry 2631
9(9082) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg
0x7fe1fd3d6ed8 with 1 -> 1
9(9082) DEBUG: auth_db [auth_db_mod.c:263]: w_is_subscriber(): uri [
sip:301@teste.demo.pt] table [subscriber] flags [1]
9(9082) DEBUG: <core> [db_res.c:116]: db_new_result(): allocate 56 bytes
for result set at 0x7fe2174742e0
9(9082) DEBUG: db_mysql [km_res.c:66]: db_mysql_get_columns(): 2 columns
returned from the query
9(9082) DEBUG: <core> [db_res.c:154]: db_allocate_columns(): allocate 16
bytes for result names at 0x7fe217474678
9(9082) DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 8
bytes for result types at 0x7fe216d82c08
9(9082) DEBUG: db_mysql [km_res.c:84]: db_mysql_get_columns(): allocate 16
bytes for RES_NAMES[0] at 0x7fe217474630
9(9082) DEBUG: db_mysql [km_res.c:91]: db_mysql_get_columns():
RES_NAMES(0x7fe217474630)[0]=[domain]
9(9082) DEBUG: db_mysql [km_res.c:135]: db_mysql_get_columns(): use
DB1_STRING result type
9(9082) DEBUG: db_mysql [km_res.c:84]: db_mysql_get_columns(): allocate 16
bytes for RES_NAMES[1] at 0x7fe2174745e8
9(9082) DEBUG: db_mysql [km_res.c:91]: db_mysql_get_columns():
RES_NAMES(0x7fe2174745e8)[1]=[rpid]
9(9082) DEBUG: db_mysql [km_res.c:99]: db_mysql_get_columns(): use DB1_INT
result type
9(9082) DEBUG: <core> [db_res.c:184]: db_allocate_rows(): allocate 16
bytes for rows at 0x7fe2174745a0
9(9082) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 64 bytes
for row values at 0x7fe217469af0
9(9082) DEBUG: <core> [db_val.c:115]: db_str2val(): converting STRING [
teste.demo.pt]
9(9082) DEBUG: <core> [db_val.c:71]: db_str2val(): converting INT [10]
9(9082) DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 2 columns
9(9082) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing
RES_NAMES[0] at 0x7fe217474630
9(9082) DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing
RES_NAMES[1] at 0x7fe2174745e8
9(9082) DEBUG: <core> [db_res.c:92]: db_free_columns(): freeing result
names at 0x7fe217474678
9(9082) DEBUG: <core> [db_res.c:97]: db_free_columns(): freeing result
types at 0x7fe216d82c08
9(9082) DEBUG: <core> [db_res.c:52]: db_free_rows(): freeing 1 rows
9(9082) DEBUG: <core> [db_row.c:95]: db_free_row(): freeing row values at
0x7fe217469af0
9(9082) DEBUG: <core> [db_res.c:60]: db_free_rows(): freeing rows at
0x7fe2174745a0
9(9082) DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set
at 0x7fe2174742e0
9(9082) DEBUG: uac [uac.c:441]: w_replace_from(): dsp=0x7fff3aad9690
(len=3) , uri=0x7fff3aad96a0 (len=29)
9(9082) DEBUG: uac [replace.c:291]: replace_uri(): removing display ["301"]
9(9082) DEBUG: uac [replace.c:331]: replace_uri(): uri to replace [
sip:301@teste.demo.pt]
9(9082) DEBUG: uac [replace.c:332]: replace_uri(): replacement uri is [
sip:301@teste.demo.pt]
9(9082) DEBUG: uac [replace.c:443]: replace_uri(): encode
is=<VnRhajNkWUhoNHZhY2tNaHFnOWhWdGFqM2RZSGg-> len=40
9(9082) DEBUG: siputils [checks.c:97]: has_totag(): no totag
9(9082) INFO: <script>: NATMANAGE: AUTOMATIC : - S=<null> - <null>
M=INVITE IP=100.64.250.254:5060 (10.0.20.103:5062)
ID=313433363433323332313432323132-2dqr4288p1lg
9(9082) DEBUG: rtpengine [rtpengine_funcs.c:140]: check_content_type():
type <application/sdp> found valid
9(9082) DEBUG: rtpengine [rtpengine.c:1527]: rtpp_function_call(): proxy
reply: d3:sdp417:v=0
o=root 1097951374 1097951374 IN IP4 100.64.250.4
s=call
c=IN IP4 100.64.250.4
t=0 0
m=audio 41660 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:41661
6:result2:oke
9(9082) DEBUG: <core> [msg_translator.c:422]: clen_builder():
content-length: 417 (417)
9(9082) DEBUG: <core> [msg_translator.c:158]: check_via_address():
(10.0.20.103, 10.0.20.103, 0)
9(9082) DEBUG: tm [t_hooks.c:266]: run_trans_callbacks_internal(): DBG:
trans=0x7fe1fd3e5c68, callback type 4194304, id 0 entered
9(9082) DEBUG: siptrace [siptrace.c:1865]: pipport2su(): the port string
is 5060
9(9082) DEBUG: siptrace [siptrace.c:1865]: pipport2su(): the port string
is 5062
9(9082) DEBUG: <core> [proxy.c:265]: mk_proxy(): doing DNS lookup...
9(9082) DEBUG: siptrace [siptrace.c:1677]: trace_send_hep_duplicate():
setting up the socket_info
9(9082) DEBUG: tm [t_funcs.c:362]: t_relay_to(): SER: new transaction
fwd'ed
9(9082) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg
0x7fe1fd3d6ed8 with 1 -> 2
9(9082) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2457 found
on entry 2631
9(9082) DEBUG: dialog [dlg_hash.c:832]: dlg_unref(): unref dlg
0x7fe1fd3d6ed8 with 1 -> 1
Thank you for your support
Regards
José Seabra
2015-05-19 14:13 GMT+01:00 José Seabra <joseseabra4(a)gmail.com>om>:
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(a)gmail.com>om>:
--
Cumprimentos
José Seabra
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(a)gmail.com>om>:
--
Cumprimentos
José Seabra
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(a)teste.itcenter.com.pt>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:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt>
--
Cumprimentos
José Seabra
--
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
;tag=oztyflbzbx
To:
<sip:401@teste.demo.pt <sip%3A401(a)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(a)lists.sip-router.org
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3220
days inactive
3272
days old
6 comments
3 participants
participants (3)
-
Daniel-Constantin Mierla -
Dragos Oancea -
José Seabra