Hi, I want to modify the following sip packet, I want to change the SDP IP address by $di, which rule could filter this packet? Something like: [ if method("OK") and $di is RFC1918 ] ??? And with which function could I change the Ip address in the SDP content? In this case I would like to change 172.16.99.2 by 190.244.125.41. Thanks in advance,
Lucas
T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP] SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060. Via: SIP/2.0/TCP 192.168.15.231:1952 ;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055. Record-Route: sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e. Record-Route: sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e. From: "Lucas"sip:1104@67.152.18.231;tag=b11c6a4e. To: sip:1101@67.152.18.231;tag=as5356a251. Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM.. CSeq: 2 INVITE. Server: Asterisk PBX 1.8.4.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: sip:1101@172.16.99.2:5080. Remote-Party-ID: "Gus Office One" <sip:1101@67.152.18.231
;party=called;privacy=off;screen=no.
Content-Type: application/sdp. Content-Length: 305. . v=0. o=root 1124623741 1124623741 IN IP4 172.16.99.2. s=Asterisk PBX 1.8.4.1. c=IN IP4 172.16.99.2. t=0 0. m=audio 13640 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
Take a look at the nathelper module: http://www.kamailio.org/docs/modules/3.1.x/modules_k/nathelper.html#id259214...
Regards, Ovidiu Sas
On Tue, Jun 14, 2011 at 6:52 PM, Lucas Alvarez lucasaa@gmail.com wrote:
Hi, I want to modify the following sip packet, I want to change the SDP IP address by $di, which rule could filter this packet? Something like: [ if method("OK") and $di is RFC1918 ] ??? And with which function could I change the Ip address in the SDP content? In this case I would like to change 172.16.99.2 by 190.244.125.41. Thanks in advance, Lucas T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP] SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060. Via: SIP/2.0/TCP 192.168.15.231:1952;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055. Record-Route: sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e. Record-Route: sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e. From: "Lucas"sip:1104@67.152.18.231;tag=b11c6a4e. To: sip:1101@67.152.18.231;tag=as5356a251. Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM.. CSeq: 2 INVITE. Server: Asterisk PBX 1.8.4.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: sip:1101@172.16.99.2:5080. Remote-Party-ID: "Gus Office One" sip:1101@67.152.18.231;party=called;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 305. . v=0. o=root 1124623741 1124623741 IN IP4 172.16.99.2. s=Asterisk PBX 1.8.4.1. c=IN IP4 172.16.99.2. t=0 0. m=audio 13640 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv.
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