Hi:<br>I wonder if this is the correct place to post a question on asterisk sip.<br>The topology of my setup is as follows:<br>Cisco 7960-----sip----Asterisk-----sip---------SipProxy.<br><br>sip.conf <br>[24057064]<br>type=friend
<br>host=dynamic ; This device needs to register<br>;nat=yes ; X-Lite is behind a NAT router<br>canreinvite=no ; Typically set to NO if behind NAT<br>context=employee
<br>;disallow=all<br>;allow=gsm ; GSM consumes far less bandwidth than ulaw<br>allow=ulaw<br>;<br>[sh-frodo]<br>type=peer<br>host=<a href="http://10.0.20.220">10.0.20.220</a><br>context=incoming_calls
<br>dtmfmode=rfc2833<br>deny=<a href="http://0.0.0.0/0">0.0.0.0/0</a><br>permit=<a href="http://10.0.20.220/32">10.0.20.220/32</a><br>allow=ulaw<br>canreinvite=yes<br><br><br>extension.conf<br>[incoming_calls]<br>exten=>24057063,1,Dial(SIP/24057063)
<br>exten=>24057064,1,Dial(SIP/24057064)<br><br>[outbound_local]<br>exten=>_800XXXXXXX,1,Dial(SIP/sh-frodo)/${EXTEN:1}<br>exten=>_800XXXXXXX,n,Congestion()<br>exten=>_800XXXXXXX,n,Hangup()<br>exten=>_386XXXXXXXX,1,Dial(SIP/sh-frodo)/${EXTEN:0}
<br>exten=>_386XXXXXXXX,n,Congestion()<br>exten=>_386XXXXXXXX,n,Hangup()<br><br>[employee]<br>include=>outbound_local<br><br>exten=>24057063,1,Dial(SIP/24057063)<br>exten=>24057064,1,Dial(SIP/24057064)<br><br>
The Invite message from cisco ip phone to asterisk is as follows:<br>INVITE <a href="mailto:sip:38624057063@10.0.49.44">sip:38624057063@10.0.49.44</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://10.74.48.41:5042">10.74.48.41:5042
</a>;branch=z9hG4bK-d87543-4058735b83308b55-1--d87543-;rport<br>Max-Forwards: 70<br>Contact: <sip:24057064@10.74.48.41:5042><br>To: "38624057063"<<a href="mailto:sip:38624057063@10.0.49.44">sip:38624057063@10.0.49.44
</a>><br>From: "24057064"<<a href="mailto:sip:24057064@10.0.49.44">sip:24057064@10.0.49.44</a>>;tag=0672ee72<br>Call-ID: M2JmMjIyMjdiZjI1Nzk1NTE0ZGVjODZhMTMzNTE3MzQ.<br>CSeq: 1 INVITE<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
<br>Content-Type: application/sdp<br>User-Agent: X-Lite release 1011s stamp 41150<br>Content-Length: 519<br><br>v=0<br>o=- 2 2 IN IP4 <a href="http://10.74.48.41">10.74.48.41</a><br>s=CounterPath X-Lite 3.0<br>c=IN IP4 <a href="http://10.74.48.41">
10.74.48.41</a><br>t=0 0<br>m=audio 24258 RTP/AVP 107 119 100 106 0 105 98 8 101<br>a=alt:1 3 : wVyQmDc6 Kv9yxoxC <a href="http://10.74.48.41">10.74.48.41</a> 24258<br>a=alt:2 2 : lrTrQrZy sn3tVDrd <a href="http://192.168.211.1">
192.168.211.1</a> 24258<br>a=alt:3 1 : KIqTV8Js uF7GKKXW <a href="http://192.168.249.1">192.168.249.1</a> 24258<br>a=fmtp:101 0-15<br>a=rtpmap:107 BV32/16000<br>a=rtpmap:119 BV32-FEC/16000<br>a=rtpmap:100 SPEEX/16000<br>a=rtpmap:106 SPEEX-FEC/16000
<br>a=rtpmap:105 SPEEX-FEC/8000<br>a=rtpmap:98 iLBC/8000<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<br><br><br>The invite message send out by Asterisk to Proxy is :<br>INVITE sip:<a href="http://10.0.20.220">10.0.20.220
</a> SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://10.0.49.44:5060">10.0.49.44:5060</a>;branch=z9hG4bK2bf1f09f;rport<br>From: "24057064" <<a href="mailto:sip:24057064@10.0.49.44">sip:24057064@10.0.49.44</a>>;tag=as65ba325a
<br>To: <sip:<a href="http://10.0.20.220">10.0.20.220</a>><br>Contact: <<a href="mailto:sip:24057064@10.0.49.44">sip:24057064@10.0.49.44</a>><br>Call-ID: <a href="mailto:7590ebc779bf26692460b6ef141f2628@10.0.49.44">
7590ebc779bf26692460b6ef141f2628@10.0.49.44</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Tue, 08 Jan 2008 02:58:22 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Supported: replaces<br>Content-Type: application/sdp<br>Content-Length: 283<br><br>v=0<br>o=root 30117 30117 IN IP4 <a href="http://10.0.49.44">10.0.49.44</a><br>s=session<br>c=IN IP4 <a href="http://10.0.49.44">10.0.49.44
</a><br>t=0 0<br>m=audio 14188 RTP/AVP 0 3 8 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:8 PCMA/8000<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><br>As you might have noticed the Request Line is incorrect: INVITE sip:<a href="http://10.0.20.220">10.0.20.220</a> SIP/2.0.<br><br>I have searched the web and found one related bug <a href="http://bugs.digium.com/view.php?id=6409">
http://bugs.digium.com/view.php?id=6409</a>, However , That does not helps.<br><br>Thank you if someone could give any indication on what is wrong with my setup.<br><br>Henry<br><br>