[Asterisk-Dev] [RFT] SIP off-server REFER support
Kevin P. Fleming
kpfleming at digium.com
Tue Apr 26 22:37:45 MST 2005
Tony (anthm) has implemented support for SIP REFERs that send calls off
of the Asterisk server that handled the initial conversation; this work
has been posted as bug #3710 in Mantis.
However, the patch is complex, and there has been one (not terribly
credible, though) report of problems with it. Given that it needs more
testing, I've added it to the patches subdirectory of CVS HEAD. Any of
you that are running CVS HEAD with multiple SIP servers, please make
some time to help test this out. After 'make update', you can add it to
your build using 'make apply PATCH=bug_3710_sip_refer.patch' and then
build as normal.
Testing it will require a SIP phone/softphone with registrations to two
different servers... place a call out one of the servers, then initiate
an attended transfer to a party on the other server. If the patch does
its job, it should just work flawlessly :-)
Report your results in bug #3710, please, I want to get this one closed
out soon, it's getting rather old...
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