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 ;-)
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