[asterisk-dev] SIP Call Transfer: Transfering phone left connected

Degasperi, Matteo Matteo.Degasperi at unitn.it
Thu Feb 4 09:22:04 CST 2010


hi, I had a problem with call transfer in this situation: asterisk 1.4.26.3 A SIP Phone A (registered on a opensips server and not with asterisk) make a call to another SIP phone B (also on opensips) and then B make an attended transfer to an external number trough an asterisk server SIP -> Asterisk -> DAHDI PRI the transfer goes well but phone B left connected. i searched for a solution and i found the issue 0015833 i applied the patch in issue 0007784 now it seems to work well, but in issue 0015833#110807 viraptor says to replace refer_call->flags[0] with refer_call->owner i don't know what is the right thing to do. and if this patch can alter the behavior in other parts of the channel. Thank you Matteo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20100204/5a381dcb/attachment.htm 


More information about the asterisk-dev mailing list