[Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515

[Lea]

Dear Adrian,

Thanks for trying to help :)

Unfortunately when I issue such command in the dialplan:
exten => 5,1,Transfer(SIP/18005551212 at VonageBlah)
   (BTW: ...my VonageTrunk is called "VonageBlah")

Then the SIP debug output of the REFER request is:

REFER sip:transit.vonage.net:5060 SIP/2.0
...snip...
Refer-To: <sip:18005551212 at VonageBlah>
...snip...

As you can see, Asterisk does not understand that I am trying to 
transfer the call to the registered peer "VonageBlah" and just 
inserts the string 18005551212 at VonageBlah verbatum into the 
Refer-To: header         :(((((

BTW:  The address (transit.vonage.net:5060) after the REFER 
keyword is also bad.  It apparently is taken from the Contact: 
header in the original INVITE request received when the incoming 
call arrives to Asterisk.  Is this in compliance with the 
RFC3515 ?

If instead, this address came from the first Record-Route: 
header in the original INVITE request, then this address would 
be good.

To tell you the truth I have no idea what the proper address 
after the REFER keyword should be and where it should come from.


Regards,
Daniel Leeds

The SIP debug output of Asterisk's original INVITE request 
received when the incoming call arrives to Asterisk is below:
----------------------------------------------------------
INVITE sip:pstn_7748712 at asterisk.aa.eu:5060;suppress-features=- 
SIP/2.0
Via: SIP/2.0/UDP sphone.vopr.vonage.net 216.115.20.41:5061
Via: SIP/2.0/UDP inbound4.vonage.net 69.59.242.97:5060
Via: SIP/2.0/UDP transit.vonage.net 
63.82.77.14:5060;branch=z9hG4bKDD2
Record-Route: <sip:pstn_7748712 at sphone.vopr.vonage.net:5061>
Record-Route: <sip:pstn_7748712 at inbound4.vonage.net:5060>
From: "unknown" <sip:transit.vonage.net>;tag=893021843
To: <sip:pstn_7748712 at inbound4.vonage.net>
Call-ID: E44DE90E-817C11DA-A2DFDF96-A870513F at 63.82.77.14
CSeq: 101 INVITE
Contact: <sip:transit.vonage.net:5060>;rtpupdated=-
...snip...



> On 2006/01/10 Adrian A adrianvoip at gmail.com wrote:
> Instead of Transfer to sphone.vopr.vonage.net, try Transfer to 
one of your Vonage peers ie.
>
> exten => 5,1,Transfer(SIP/18005551212 at vonageTrunk)


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