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
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
