Hi
i generate a call from the dialplan in this mode:
exten => 1002,1,Answer()
exten => 1002,2,Dial(SIP/[EMAIL PROTECTED])
the call is generated, but after some seconds it is interrupted, here
the asterisk log:
*CLI> -- Executing Answer("SIP/host1-0819d0d0", "") in new stack
-- Executing Dial("SIP/host1-0819d0d0", "SIP/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/host-081a2610 is ringing
-- SIP/host-081a2610 answered SIP/host1-0819d0d0
-- Attempting native bridge of SIP/host1-0819d0d0 and SIP/host-081a2610
== Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0'
i've enabled sip debug, but nothing interesing has been showed
host1 is an SJphone and host is a software that implements SIP protocol.
Can you help me to guess where is the problem?
if i try to create a call from SJphone 2 SJphone all works fine.
Is possible that exists a problem in asterisk ?
where ? how can i find it ?
thanks to all
--
/*************/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/nikstresser
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