hey all,
am wondering if anyone has successfuly done a SIP attended transfer using
the REFER method (after an INVITE obviously) and the Replaces: header.
we're writing our own SIP UAC and the asterisk code seems to support it,
but we're not really sure if this is so.
we plan on the following call flows:
1. incoming call from exten 1111 is sent to our UAC with Dial()
2. our UAC makes another call (via asterisk) to exten 2222
3. 2222 answers the call
4. our UAC sends REFER with Replaces: and Call-Id: of call from 1111 to the
SIP session with asterisk for 2222.
5. asterisk bridges 1111 and 2222.
is this the way it's supposed to work ?
(am not sure if this is a -users or -dev question, so pardon the x-posting)
--
Regards, /\_/\ "All dogs go to heaven."
[EMAIL PROTECTED] (0 0) http://www.alphaque.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a $b." |
| done; done |
+=========================================================================+
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