Hi,
I've recently discovered a scenario that causes asterisk to send SIP
messages with the Request URI missing and the TO URI missing.
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.
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.
Has anyone got any idea how to stop this happening?
The SIP messages are as follows:
NOTIFY sip: SIP/2.0
Via: SIP/2.0/UDP 192.168.2.195:5062;rport;branch=z9hG4bK17f004f7
To: <sip:>
From: "David" <sip:219@192.168.2.195>;tag=as51f54c64
Call-ID: 11cd8bc246bd1cb0@192.168.2.132
CSeq: 102 NOTIFY
Contact: <sip:219@192.168.2.195:5062>
User-Agent: PBX Gateway
Event: refer;id=41590
Content-Type: message/sipfrag; version=2.0
Content-Length: 14
Subscription-state: terminated;reason=noresource
SIP/2.0 200 OK
BYE sip: SIP/2.0
Via: SIP/2.0/UDP 192.168.2.195:5062;branch=z9hG4bK4d88ed51
To: "David" <sip:219@192.168.2.195>;tag=2da39d99e5d753cd
From: <sip:839219@192.168.2.195>;tag=as51f54c64
Call-ID: 11cd8bc246bd1cb0@192.168.2.132
CSeq: 103 BYE
Route: <sip:219@192.168.2.132>
Contact: <sip:219@192.168.2.195:5062>
User-Agent: PBX Gateway
Content-Length: 0
Thank you.
Jonathan
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