[asterisk-dev] call transfer to asterik.. asterisk as an end point

[Zahid Mehmood]

Hello All.

     I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience.



I want to use asterisk for call park/pickup and have configured openser
to relay calls made to  ruri 700-720 to asterisk running on
localhost:5069





Call flow:



phone A calls  phone B  (both phones are polycom)



Phone B answers



then phone b user presses transfer and dials 700



asterisk plays back 701 as the parking lot location



phone B user presses transfer again.

   at this time phone b is not disconnected from asterisk system

   phone A is also connected to asterisk and hears 702 as the parking
lot location (as if asterisk places the user at priority 1 for that

context)





 From phone C calling 702 will connect phone C to phone A.





This was a specific example but this transfer problem is not limited to
call park only. It happens any time asterisk is the second party called
in call transfer.



Thanks in advance for your help.



--

Zahid
 





On May 8, 2007, at 1:56 PM, Christian Schlatter wrote:



I think I found out why this doesn't work as expected. After phone 1 receives REFER from phone 2, it sends a new INVITE to the asterisk server. This INVITE includes a Replaces: header that tells the receiver (asterisk) to replace an existing SIP dialog with the new one.

RFC 3891 "The SIP Replaces Header", Section 3 "UAS Behavior", defines:

"the UA attempts to accept the new INVITE, reassign the user interface and other resources of the matched dialog to the new INVITE, and shut down the replaced dialog."

But your SIP trace shows that asterisk doesn't shut down the replaced dialog (by sending a BYE), which is the reason why phone 2 does not get disconnected after hitting "transfer" the second time.


Instead of creating a new call park slot (702) when phone 1 sends the Replaces: INVITE to asterisk, asterisk should be intelligent enough to figure out that this INVITE actually replaces the existing SIP dialog with phone 2. And asterisk should not create a new park slot 702 but directly put phone 1 on hold at park slot 701 and send a BYE to phone 2.

Although asterisk supports the Replaces: header when used e.g. as a gateway, I have some doubts that the call park/pickup implementation does so too. Especially since it was designed to be used in "PBX mode" where asterisk acts as B2BUA for all involved call legs.

Maybe this should be opened as a new feature/bug request on the asterisk bug tracker. Or maybe there is a asterisk setting that controls this behavior, I'm not really an asterisk expert myself ;-)

-- 
"The fact that an opinion has been widely held is no evidence that it is not utterly absurd; indeed, in view of the silliness of the majority of mankind, a widespread belief is more often likely to be foolish than sensible." -Bertrand Russell





 
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