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