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<div dir="ltr"><font color="#000000" size="2" face="Tahoma">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.
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<p>Received from udp:10.1.10.80:5060 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" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20<br>
To: <sip:9513261429@ser.gowireless.net;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:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes):</p>
<p class="text4">ACK sip:vm9513261429@10.1.10.83:5060 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" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20<br>
t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f<br>
i: 3c26755bf15c-9iq08xqqblo6<br>
CSeq: 4 ACK<br>
Max-Forwards: 70<br>
m: <sip:91421@10.30.0.64:5060>;reg-id=1<br>
l: 0<br>
<br>
</p>
<hr>
<p>Received from udp:10.1.10.80:5060 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" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8<br>
To: <sip:9513261429@ser.gowireless.net;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: <sip:9513261429@10.1.10.83:5060><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:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes):</p>
<p class="text4">ACK sip:9513261429@10.1.10.83:5060 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" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8<br>
t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96<br>
i: 3c2676547a8d-2t5yi6jok1sv<br>
CSeq: 2 ACK<br>
Max-Forwards: 70<br>
m: <sip:91421@10.30.0.64:5060>;reg-id=1<br>
l: 0<br>
<br>
</p>
<hr>
<p>Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes):</p>
<p class="text4">BYE sip:91421@10.30.0.64:5060 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: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96<br>
To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;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>
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<div class="PlainText">Regards,<br>
<br>
Rowell</div>
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