[Kamailio-Users] CANCEL over nat - sr-users

1 Mar 2009

i have async rtp proxy setup on multihomed host.

config for transfer cancel and invites:
if(lookup("location")){
                log("found loc");
                fix_nated_contact();
                $avp(i:11)=$fU;
                $var(fr)="sip:"+$avp(i:11)+"@69.70.173.195";
                xlog("replace $var(fr)");
                if (method == "BYE" || method == "CANCEL"){
                  unforce_rtp_proxy();
#                   force_rtp_proxy("AIEOC");
                }else{
                 uac_replace_from("$avp(i:11)","$var(fr)");
                 if(force_rtp_proxy("FAIEOC"))
                         t_on_reply("2");
                }
                route(1);
                exit();
          }

cancel is routed ok, ATA get cancel in form of (ip changed):

RECEIVING FROM: 69.70.xxx.xx:5060
CANCEL sip:34249@193.110.78.12:8484 SIP/2.0
Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0
From: "Unknown" <sip:Unknown@69.70.173.195
&lt;sip%3AUnknown(a)69.70.173.195&gt;
...
 ;tag=as704e6b0d Call-ID:
735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180
To: <sip:34249@192.168.2.170 &lt;sip%3A34249(a)192.168.2.170&gt;&gt;
CSeq: 102 CANCEL
Max-Forwards: 70
asterisk
Content-Length: 0

but ignore it. same with any other ATA type(including softphones)
can u please explaine me what is incorrect in CANCEL?
cancel produced by asterisk.CALL-id is correct. here is corespondoing invite
16:06:56.0
RECEIVING FROM: 69.70.173.195:5060
INVITE sip:34249@193.110.78.12:8484 SIP/2.0
Record-Route:
<sip:69.70.173.195;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ-->
Record-Route:
<sip:192.168.2.170;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ-->
Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0
Via: SIP/2.0/UDP 192.168.1.180:5060;rport=5060;branch=z9hG4bK49fa4d6d
From: "Unknown" <sip:Unknown@69.70.173.195
&lt;sip%3AUnknown(a)69.70.173.195&gt;
...
 ;tag=as704e6b0d To: <sip:34249@192.168.2.170
&lt;sip%3A34249(a)192.168.2.170&gt;&gt;
Contact: <sip:Unknown@192.168.1.180:5060>
Call-ID: 735a38dd3d77f4f20f29f28b335f11f4(a)192.168.1.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Sun, 01 Mar 2009 14:05:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 447

v=0
o=root 8300 8300 IN IP4 69.70.173.195
s=session
c=IN IP4 69.70.173.195
t=0 0
m=audio 10336 RTP/AVP 0 8 110 97 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=nortpproxy:yes

-- 
Merkulov Alexander