Hello users, i am working on directly calling the numbers from the sip provider of my choice from asterisk using Dial command as follows.
extensions.conf [dial-out] exten => _XXXXXXXXXX,1,NoOp(Dialing out) exten => _XXXXXXXXXX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsp...@host:port , 20,r) exten => _XXXXXXXXXX,n,Hangup() //so i am trying to call the number using voip provider details i have but i am getting the following error in asterisk CLI SIP/408XXXXXXX:xxxxx::XXXXXXX:u...@xxxxxx Called 140XXXXXXXX:xxxxx::XXXXXXX:u...@xxxxxx -- SIP/xxxxxx-0a155070 is circuit-busy when i try with other service provider i am getting a similar error in asterisk CLI SIP/1408XXXXXXXXX:yyyyy::YYYYYY:u...@yyyyyyyyyyy Got SIP response 500 "Nice try" back from 64.xx.xx.xx -- SIP/yyyyyyyyyyy-0a16ac20 is circuit-busy my idea is to allow users to enter their own voip providers for outgoing calls so that customer can use his own voip provider i am NOT LOOKING FOR A SOLUTION in /etc/sip.conf entries like register => username:passw...@myprovider [myprovider] username= secret= fromuser= fromdomain= host= any help is appreciated. Thanks srinvias
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