i have async rtp proxy setup on multihomed host.<br><br>config for transfer cancel and invites:<br>if(lookup("location")){<br> log("found loc");<br> fix_nated_contact();<br>
$avp(i:11)=$fU;<br> $var(fr)="sip:"+$avp(i:11)+"@<a href="http://69.70.173.195">69.70.173.195</a>";<br> xlog("replace $var(fr)");<br> if (method == "BYE" || method == "CANCEL"){<br>
unforce_rtp_proxy();<br># force_rtp_proxy("AIEOC");<br> }else{<br> uac_replace_from("$avp(i:11)","$var(fr)"); <br> if(force_rtp_proxy("FAIEOC"))<br>
t_on_reply("2");<br> }<br> route(1);<br> exit();<br> }<br clear="all"><br>cancel is routed ok, ATA get cancel in form of (ip changed):<br>
<br><br>RECEIVING FROM: 69.70.xxx.xx:5060<br>CANCEL <a href="http://sip:34249@193.110.78.12:8484">sip:34249@193.110.78.12:8484</a> SIP/2.0<br>Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0<br>From: "Unknown" <<a href="mailto:sip%3AUnknown@69.70.173.195">sip:Unknown@69.70.173.195</a>>;tag=as704e6b0d<br>
Call-ID: <a href="mailto:735a38dd3d77f4f20f29f28b335f11f4@192.168.1.180">735a38dd3d77f4f20f29f28b335f11f4@192.168.1.180</a><br>To: <<a href="mailto:sip%3A34249@192.168.2.170">sip:34249@192.168.2.170</a>><br>CSeq: 102 CANCEL<br>
Max-Forwards: 70<br>asterisk<br>Content-Length: 0<br><br>but ignore it. same with any other ATA type(including softphones)<br>can u please explaine me what is incorrect in CANCEL?<br>cancel produced by asterisk.CALL-id is correct. here is corespondoing invite<br>
16:06:56.0 <br>RECEIVING FROM: <a href="http://69.70.173.195:5060">69.70.173.195:5060</a><br>INVITE <a href="http://sip:34249@193.110.78.12:8484">sip:34249@193.110.78.12:8484</a> SIP/2.0<br>Record-Route: <sip:69.70.173.195;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--><br>
Record-Route: <sip:192.168.2.170;r2=on;lr=on;ftag=as704e6b0d;last_from=bWVyYWxtZXJhbG1ldWFwdGRqaHVvZXJgaQ--><br>Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK6e35.c727f225.0<br>Via: SIP/2.0/UDP 192.168.1.180:5060;rport=5060;branch=z9hG4bK49fa4d6d<br>
From: "Unknown" <<a href="mailto:sip%3AUnknown@69.70.173.195">sip:Unknown@69.70.173.195</a>>;tag=as704e6b0d<br>To: <<a href="mailto:sip%3A34249@192.168.2.170">sip:34249@192.168.2.170</a>><br>Contact: <<a href="http://sip:Unknown@192.168.1.180:5060">sip:Unknown@192.168.1.180:5060</a>><br>
Call-ID: <a href="mailto:735a38dd3d77f4f20f29f28b335f11f4@192.168.1.180">735a38dd3d77f4f20f29f28b335f11f4@192.168.1.180</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 69<br>Date: Sun, 01 Mar 2009 14:05:34 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Type: application/sdp<br>Content-Length: 447<br><br>v=0<br>o=root 8300 8300 IN IP4 69.70.173.195<br>s=session<br>c=IN IP4 69.70.173.195<br>
t=0 0<br>m=audio 10336 RTP/AVP 0 8 110 97 18 4 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:110 speex/8000<br>a=rtpmap:97 iLBC/8000<br>a=fmtp:97 mode=30<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>
a=rtpmap:4 G723/8000<br>a=fmtp:4 annexa=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br>a=nortpproxy:yes<br><br><br><br>-- <br>Merkulov Alexander<br>