<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; white-space: pre-wrap; ">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</span></body></html>