asterisk users - Jul 2012 - sip and extensions

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:funnytiger123 at 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()
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120705/0a9c8ae1/attachment.htm>

----- Original Message ----- 
> I am new. Here is the code that I am playing with on CentOS 6.x
> register => 5552530146:funnytiger123 at 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

*PLEASE* if that is your real username/password with your VoIP provider change
it immediately. Just FYI, you've broadcast it to (tens or hundreds of)
thousands of list readers. I have to believe some are of the nefarious type that
would love to use your account for free calling at your expense. :/

--Tim

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:funnytiger123 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20120706/7e5fd34b/attachment.htm>