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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060102/1e45fa86/attachment.htm
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