2010/10/25 Nauman Sulaiman <[email protected]>:
> Hi, i have examined some SIP traces of attended transfers using some PBX's
> such as Asterisk and 3CX using X-Lite softphones and am finding that there
> are no Replaces headers in the Invites to the transfer target. So where is
> the Replaces header in Invites used these days and how could i set up some
> call flow to observe it?
I expect you mean the "call capture" feature, so a phone can answer a
call ringing in other device. RFC 4235 ("dialog" event) and RFC 3891
("Replaces" header) define this mechanism in this way:
- Phone A subscribes to "dialog" event of phone B so it receives
notifications about new calls from/to phone B.
- These notifications contains full information about the dialog
(call-id, from-tag) so Phone A can generate an INVITE with Replaces
header, send it to the caller and capture the call.
However Asterisk implementation of this mechanism is a joke:
- First of all, asterisk supports subscription to "dialog" event but
in a custom way (notifications don't contain info about dialog's
call-id and from-tag, just the caller and called uri's).
- Also, Asterisk doesn't understand receiving an INVITE with Replaces header.
- Instead, phone B must generate an INVITE in which the RURI is a
special extension defined in asterisk (i.e. *8 +
EXTENSION_TO_CAPTURE).
- So phone B sends an "INVITE sip:*8 + PHONE_A_EXTEN" to Asterisk, and
Asterisk finds the *first* (or any other) call in which phone A is
involved, and connects such call to phone B.
Of course, this last step sucks as in case A is receiving various
calls, there is no way to tell asterisk which call phone B wants to
capture.
So, Asterisk implementation of this mechanism is a private and painful
specification. And the worst is the fact that some vendors (as
Cisco/Linksys) implement this NON-standard specification in their
phones (called "Asterisk BLF"). This is a pain.
--
Iñaki Baz Castillo
<[email protected]>
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