Hello, If you would like to make out bound call (from Asterisk to SIP provider), it is fine.
But if you want have inbound call (from SIP provider to Asterisk). I think you are supposed to have something like this sip.conf register => 5552530146:<your_password>@sip3.voipvoip.com/5552530146 [5552530146] ....... context=incoming extensions.conf [incoming] ;first creating extensions for your local users exten => 5552530146,1,Goto(5552530146_incomming,s,1) [5552530146_incomming] ;more logic wish it would help. On Jul 5, 2012, at 11:44 PM, Thomas Perron wrote: > I am new. Here is the code that I am playing with on CentOS 6.x > > When I dial the number that corresponds w/ my SIP account I get a recording: > "reached a non-working number........" > > I built Asterisk a few times last year and am now back working on a similar > project. In my view, there is something wrong in sip.conf > I don't remember using a file that long to get a basic call set up. The > format was provided to me by voipvoip.com (the SIP provider). > > Does anyone have any comments please? I just want a very simple config to > get my machine to recognize a call to the SIP provider. > > Here is results of sip show registry: > > Host dnsmgr Username Refresh State > Reg.Time > sip3.voipvoip.com:5060 N 5552530146 285 > Registered Thu, 05 Jul 2012 21:39:56 > 1 SIP registrations. > > Here is sip and extensions.conf > > sip.conf > > [general] > register => 5552530146:funnytiger...@sip3.voipvoip.com > ; > > [sip3.voipvoip.com] > > [outgoing] > username=5552530146 > type=peer > qualify=yes > secret=funnytiger123 > nat=auto > insecure=very > host=69.90.209.57 > fromuser=5552530146 > fromdomain=69.90.209.57 > dtmfmode=rfc2833 > allow=g729 > allow=ilbc > allow=ulaw > allow=alaw > disallow=all > srvlookup=no > > [incoming] > username=5552530146 > type=user > secret=funnytiger123 > nat=auto > insecure=very > host=69.90.209.57 > fromdomain=69.90.209.57 > dtmfmode=rfc2833 > context=incoming > allow=g729 > allow=ulaw > allow=alaw > allow=ilbc > disallow=all > srvlookup=no > > > > extensions.conf > > [general] > > ; > ; > [incoming] > ;first creating extensions for your local users > exten=> s,1,Dial(SIP/17037175555) > exten=> s,2,Hangup() > > > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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