hi all,
i am trying to execute a call file in asterisk by placing it in the
outgoing directory.In my system i m running rtpproxy and openser also.Asterisk
is communicating with openser because i am able to make incoming calls to
asterisk.But when i try to put call file in the outgoing directory we are
getting the following errors.
Attempting call on SIP/104@192.168.0.111 for application
Playback(demo-congrats) (Retry 1)
-- Got SIP response 482 "Loop Detected" back from 192.168.0.111
Jan 2 16:58:51 NOTICE[4685]: pbx_spool.c:266 attempt_thread: Call failed to go
through, reason 8
Can anyone help me regarding this please??
with regards
vicky
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Hello,
as You are running two processes handling SIP (asterisk and openser), I
think the Call-File addresses the wrong instance.
If Your callfile contains a line like
Channel: SIP/accountname
try something like
Channel: SIP/accountname@ipaddress:port
where ipaddress and port addressing the responable instance.
HTH,
Karsten
hi, thanks for reply. even after specifying the port, we are getting the same error. with regards vicky On Mon, 02 Jan 2006 Karsten Wemheuer wrote :>Hello, > >as You are running two processes handling SIP (asterisk and openser), I >think the Call-File addresses the wrong instance. > >If Your callfile contains a line like > Channel: SIP/accountname >try something like > Channel: SIP/accountname@ipaddress:port >where ipaddress and port addressing the responable instance. > >HTH, > >Karsten > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060102/71661d7b/attachment.htm