[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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