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<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>I've recently
discovered a scenario that causes asterisk to send SIP messages with the Request
URI missing and the TO URI missing.</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>It happens when a
call goes out over a Zap channel from an internal SIP phone.
When the internal SIP phone initiates a transfer to another SIP phone the
transfer takes place but the NOTIFY and BYE message sent by asterisk to the
first SIP phone are missing the request URI and the NOTIFY is also missing the
TO header URI.</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>The result is that
the initiator of the transfer does not receive confirmation that the transfer as
taken place and still thinks it is in the call.</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>Has anyone got any
idea how to stop this happening?</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>The SIP messages are
as follows:</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>NOTIFY sip:
SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.2.195:5062;rport;branch=z9hG4bK17f004f7<BR>To: <sip:><BR>From:
"David" <sip:219@192.168.2.195>;tag=as51f54c64<BR>Call-ID: <A
href="mailto:11cd8bc246bd1cb0@192.168.2.132">11cd8bc246bd1cb0@192.168.2.132</A><BR>CSeq:
102 NOTIFY<BR>Contact: <sip:219@192.168.2.195:5062><BR>User-Agent: PBX
Gateway<BR>Event: refer;id=41590<BR>Content-Type: message/sipfrag;
version=2.0<BR>Content-Length: 14<BR>Subscription-state:
terminated;reason=noresource</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>SIP/2.0 200
OK</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>BYE sip:
SIP/2.0<BR>Via: SIP/2.0/UDP 192.168.2.195:5062;branch=z9hG4bK4d88ed51<BR>To:
"David" <sip:219@192.168.2.195>;tag=2da39d99e5d753cd<BR>From:
<sip:839219@192.168.2.195>;tag=as51f54c64<BR>Call-ID: <A
href="mailto:11cd8bc246bd1cb0@192.168.2.132">11cd8bc246bd1cb0@192.168.2.132</A><BR>CSeq:
103 BYE<BR>Route: <sip:219@192.168.2.132><BR>Contact:
<sip:219@192.168.2.195:5062><BR>User-Agent: PBX Gateway<BR>Content-Length:
0</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial size=2>Thank
you.</FONT></SPAN></DIV>
<DIV><SPAN class=804171708-07102004></SPAN> </DIV>
<DIV><SPAN class=804171708-07102004><FONT face=Arial
size=2>Jonathan</FONT></SPAN></DIV></BODY></HTML>