Hi
We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected
to old PBX, and some SIP phones, used by a callcenter with queues.
Almost all calls are incoming (through E1 line), answered by some
callcenter operator (using SIP phones, call assigned by queue app),
and in some cases, are transferred to some other extension on the old
PBX or other SIP.
We had problems with Music on Hold (on the queue) and with transfers
on version 1.0.3. We now upgraded to 1.0.7 and the MoH problem is
gone, but we still have some transfer problems.
What happens is that sometimes when one callcenter op (SIP client)
does a transfer to another SIP or an extension that is mapped to a FXO
line (old PBX), we get a half-call: the caller hears the called
station, but the called station (the one the call is transferred to)
does not here the caller.
As we need attended transfer, the calls are made from the SIP phone
(Xten), using the transfer button (not blind transfers).
Don't really know how to debug this. Is there a log I can see that can
help me pinpoint the problem?. On that log, what should we be looking
for? I'm used to debug this kind of problems in general, but are not
familiar with SIP protocol nor Asterisk debugging.
We tried to change SIP phones, but its the same. Note that it happens
with calls that have one end on the E1 and the other to FXO, both
local to Asterisk ("joined" by a SIP phone), so it does not seems to
be a codec problem.
Thanks for any advice.
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