Hi,
I'm having problems with sip remote attended transfer using 2 asterisk
boxes (same version, latest 1.4.X). Whenever I transfer from a call
from box A to a call on box B, one call leg of the transferring phone
is not disconnected (the one that is normally dropped by server side,
phone disconnects the other one). The same situation works perfectly
with local attended transfer.
Is anyone using remote attended transfers with asterisk? Does it work
for you? Do you use any workarounds? I'm asking here, because it would
be strange if that functionality was broken since 1.4.8 and noone
noticed ;)
Exact scenario I'm using is described in the bug:
https://issues.asterisk.org/view.php?id=15833
Thanks for any help.
--
Kind regards,
Stanis?aw Pitucha, Gradwell Voip Engineer
T: 01225 800 831 | F: 01225 800 801 | E: stan at gradwell.net | www.gradwell.com
Gradwell ? Internet for Business People
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