[asterisk-users] Issues with making Transfers

[Dan Brummer]

Asterisk 1.2.9.1 is the version I'm on.

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan
Brummer
Sent: Tuesday, July 11, 2006 8:30 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Issues with making Transfers


Hello,
I am having a problem with transferring calls that come in from the
outside.  Users have been calling in to the PRI that's on the Cisco GW,
then they are passed into Asterisk via SIP and to the end phone (Polycom
501/601) using SIP.  When that user tries to transfer that call to
another extension, the call disconnects and hangs in the air and doesn't
do anything.  The call shows active in the Cisco GW but no where to be
found in asterisk.  Here is some log output of a transfer attempt:
 
    -- Stopped music on hold on SIP/10.25.118.2-b7b4e520
  == Spawn extension (ANC, 4023, 2) exited non-zero on
'SIP/4023-ebbf<ZOMBIE>'
    -- SIP/2198-3780 answered SIP/10.25.118.2-b7b4e520
    -- Attempting native bridge of SIP/10.25.118.2-b7b4e520 and
SIP/2198-3780
    -- Incoming call: Got SIP response 500 "Internal Server Error" back
from 10.45.25.12
 
 
I'm not sure if the SIP 500 error is relative to my issue.  Any ideas on
what could be causing SIP transfers to hang or drop?
 
Thank you,
Dan
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