<div>below is the transaction of the failed mediaproxy invite. I allready could tell that replies go through openser, but I don't see the reason why ser doesn't see them as replies (and use the mediaproxy function).</div>
<div> </div>
<div>as you can see, the invite from <ip client> to <ip asterisk> (through <ip OPENSER>, which is also ip of mediaproxy) goes in one direction good (the ip in the SDP is changed from <ip client> to <ip openser>, but the return path en the OK (with it's SDP) is not changed
</div>
<div> </div>
<div>I did a tcpdump with a call between 2 clients, where the proxy works, and the only difference I see is that in the reply of asterisk, there is no rinstance field in the contact header</div>
<div> </div>
<div>thanks</div>
<div>arne</div>
<div> </div>
<div>U <ip client>:5060 -> <ip OPENSER>:5060<br> INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<
sip:701@sipgat<br> <a href="http://e.evonet.be">e.evonet.be</a>>..Call-ID: <a href="mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip">1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip</a> client>..CSeq: 1 INVITE..Via: SIP/2.0/UDP <ip client>:5060;rport;branch=z9hG4bK-7a70a-1d
<br> e331c2-69dc..Max-Forwards: 70..Supported: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP1<br> 0 SP v1.0.1 (Build 3) 3.0.5.1..Allow-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip client>:5060;transport=UDP>..Session-Expires: 1800..Content-
<br> Type: application/sdp..Content-Length: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip client>..t=0 0..m=audio 50000 RTP/AVP 18 0 8..<br> a=fmtp:18 annexb=yes..a=ptime:40..a=SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv..
<br>#<br> </div>
<div>U <ip OPENSER>:5060 -> <ip asterisk>:5060<br> INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..Record-Route: <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne" <
sip:1002@si<br> <a href="http://pgate.evonet.be">pgate.evonet.be</a>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: <a href="mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip">
1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip</a> client>..C<br> Seq: 1 INVITE..Via: SIP/2.0/UDP <ip OPENSER>;branch=0..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Max-Forwards: 69..Supp
<br> orted: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP10 SP v1.0.1 (Build 3) 3.0.5.1..Allo<br> w-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip client>:5060;transport=UDP>..Session-Expires: 1800..Content-Type: application/sdp..Content-Leng
<br> th: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip OPENSER>..t=0 0..m=audio 60106 RTP/AVP 18 0 8..a=fmtp:18 annexb=yes..a=ptime:40..a<br> =SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv..
<br>#<br> </div>
<div>U <ip asterisk>:5060 -> <ip OPENSER>:5060<br> SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-
<br> 69dc..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70<br> <a href="mailto:a-1de331be-529@<ip">
a-1de331be-529@<ip</a> client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@<br> <ip asterisk>>..Content-Length: 0....
<br>#<br> </div>
<div>U <ip OPENSER>:5060 -> <ip client>:5060<br> SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c
<br> 4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: <a href="mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip">1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip</a> client>..CSeq: 1 INVITE..User-Agent: Asteri
<br> sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Length: 0....<br>#<br> </div>
<div>U <ip asterisk>:5060 -> <ip OPENSER>:5060<br> SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc
<br> ..Record-Route: <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f<br> ..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID:
<a href="mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip">1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip</a> client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX<br> ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: application/sdp..Content-Length: 188....v=
<br> 0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4 <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=<br> silenceSupp:off - - - -..<br>#<br> </div>
<div>U <ip OPENSER>:5060 -> <ip client>:5060<br> SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Record-Route: <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70
<br> a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID:<br> <a href="mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip">
1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip</a> client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NO<br> TIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: application/sdp..Content-Length: 188....v=0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4
<br> <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=silenceSupp:off - - - -..<br>#<br> </div>
<div> </div>
<div><br><br> </div>
<div><span class="gmail_quote">2006/9/21, Daniel-Constantin Mierla <<a href="mailto:daniel@voice-system.ro">daniel@voice-system.ro</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hello,<br><br>watch the network traffic with ngrep on your sip server. You can see the<br>call flow which may help to identify the issue. You can paste it to the
<br>list and someone may give you hints.<br><br>Cheers,<br>Daniel<br><br><br>On 09/21/06 12:28, Arne Van Theemsche wrote:<br>> hi<br>><br>> my users subscribe with openser, en asterisk is used as connectivity<br>
> to pstn<br>><br>> i am now installing a mediaproxy, for all users, so every call goes<br>> via a mediaproxy.<br>><br>> I'm doing this as follows (relevant statements only)<br>><br>> in route<br>>
<br>> #I installed the t_on_reply here to be sure that every reply<br>> gets parsed, but normally in the INVITE section should be enough?<br>> t_on_reply("1");<br>><br>> if (method==INVITE) {
<br>> use_media_proxy();<br>> }<br>><br>><br>> onreply_route[1] {<br>> log(-3,"reply received");<br>> if (!search("^Content-Length:[ ]*0")) {
<br>> log(-3,"using mediaproxy");<br>> use_media_proxy();<br>> };<br>> }<br>><br>><br>> the weird is, for all local users, this works fine, but as soon as
<br>> asterisk is involved, the reply doesn't get triggered (not seeing the<br>> "reply received" either, only when disconnecting the call). The call<br>> get's established fine, asterisk is sending media to the mediaproxy,
<br>> but the SDP towards the calling phone is not modified (since the<br>> onreply isn't triggered)<br>><br>> am I missing something here?<br>><br>> thanks<br>> Arne<br>><br>> ------------------------------------------------------------------------
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