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<div dir="ltr"><font face="Tahoma" size="2" color="#000000">Hi,</font></div>
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<div dir="ltr"><font face="tahoma">We are having issues where the "OK" or "ACK" is
that is coming from the phone is not relayed by OpenSER to
Asterisk.</font></div>
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<div dir="ltr"><font face="tahoma">Below is the sip trace... I am also attaching a
tcpdump. Please help what we can do. </font></div>
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<div dir="ltr"><font face="tahoma">
<p>Received from udp:<a href="http://10.1.10.80:5060">10.1.10.80:5060</a> at 26/10/2011 10:22:41:476 (490 bytes):</p>
<p class="text4">SIP/2.0 481 Call/Transaction Does Not Exist<br>Via: SIP/2.0/UDP
10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060<br>From:
"Virgil Menendez" <<a href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=6wkdms1r20<br>To:
<<a href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as0b87218f<br>Call-ID:
3c26755bf15c-9iq08xqqblo6<br>CSeq: 4 INVITE<br>Server: Asterisk PBX
1.8.7.1<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH<br>Supported: replaces, timer<br>Content-Length: 0<br><br></p>
<hr>
<p>Sent to udp:<a href="http://10.1.10.80:5060">10.1.10.80:5060</a> at 26/10/2011 10:22:41:481 (387 bytes):</p>
<p class="text4">ACK <a href="http://sip:vm9513261429@10.1.10.83:5060">sip:vm9513261429@10.1.10.83:5060</a> SIP/2.0<br>v: SIP/2.0/UDP
10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport<br>Route:
<sip:10.1.10.80;lr=on><br>f: "Virgil Menendez"
<<a href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=6wkdms1r20<br>t:
<<a href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as0b87218f<br>i:
3c26755bf15c-9iq08xqqblo6<br>CSeq: 4 ACK<br>Max-Forwards: 70<br>m:
<<a href="http://sip:91421@10.30.0.64:5060">sip:91421@10.30.0.64:5060</a>>;reg-id=1<br>l: 0<br><br></p>
<hr>
<p>Received from udp:<a href="http://10.1.10.80:5060">10.1.10.80:5060</a> at 26/10/2011 10:22:42:130 (868 bytes):</p>
<p class="text4">SIP/2.0 200 OK<br>Via: SIP/2.0/UDP
10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060<br>Record-Route:
<sip:10.1.10.80;lr=on><br>From: "Virgil Menendez"
<<a href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=qi3i8ze6z8<br>To:
<<a href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as3f8c0f96<br>Call-ID:
3c2676547a8d-2t5yi6jok1sv<br>CSeq: 2 INVITE<br>Server: Asterisk PBX
1.8.7.1<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH<br>Supported: replaces, timer<br>Session-Expires:
1800;refresher=uas<br>Contact:
<<a href="http://sip:9513261429@10.1.10.83:5060">sip:9513261429@10.1.10.83:5060</a>><br>Content-Type:
application/sdp<br>Content-Length: 256<br><br>v=0<br>o=root 1355451627
1355451627 IN IP4 10.1.10.83<br>s=Asterisk PBX 1.8.7.1<br>c=IN IP4
10.1.10.83<br>t=0 0<br>m=audio 16094 RTP/AVP 0 8 101<br>a=rtpmap:0
PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101
telephone-event/8000<br>a=fmtp:101 0-16<br>a=ptime:20<br>a=sendrecv<br></p>
<hr>
<p>Sent to udp:<a href="http://10.1.10.80:5060">10.1.10.80:5060</a> at 26/10/2011 10:22:42:132 (385 bytes):</p>
<p class="text4">ACK <a href="http://sip:9513261429@10.1.10.83:5060">sip:9513261429@10.1.10.83:5060</a> SIP/2.0<br>v: SIP/2.0/UDP
10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport<br>Route:
<sip:10.1.10.80;lr=on><br>f: "Virgil Menendez"
<<a href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=qi3i8ze6z8<br>t:
<<a href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as3f8c0f96<br>i:
3c2676547a8d-2t5yi6jok1sv<br>CSeq: 2 ACK<br>Max-Forwards: 70<br>m:
<<a href="http://sip:91421@10.30.0.64:5060">sip:91421@10.30.0.64:5060</a>>;reg-id=1<br>l: 0<br><br></p>
<hr>
<p>Received from udp:<a href="http://10.1.10.80:5060">10.1.10.80:5060</a> at 26/10/2011 10:22:42:232 (503 bytes):</p>
<p class="text4">BYE <a href="http://sip:91421@10.30.0.64:5060">sip:91421@10.30.0.64:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP
10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0<br>Via: SIP/2.0/UDP
10.1.10.83:5060;branch=z9hG4bK69f53cf1<br>Max-Forwards: 69<br>From:
<<a href="mailto:sip%3A9513261429@ser.gowireless.net">sip:9513261429@ser.gowireless.net</a>;user=phone>;tag=as3f8c0f96<br>To:
"Virgil Menendez"
<<a href="mailto:sip%3A91421@ser.gowireless.net">sip:91421@ser.gowireless.net</a>>;tag=qi3i8ze6z8<br>Call-ID:
3c2676547a8d-2t5yi6jok1sv<br>CSeq: 102 BYE<br>User-Agent: Asterisk PBX
1.8.7.1<br><strong>X-Asterisk-HangupCause: Protocol error,
unspecified<br></strong>X-Asterisk-HangupCauseCode: 111<br>Content-Length:
0<br></p></font></div>
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<div class="PlainText">Regards,<br><br>Rowell</div></font></div></div>