[asterisk-users] SIP Woes

[Joshua Colp]

----- Original Message -----
From: Dave Hope
[mailto:dave at davehope.co.uk]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com]
Sent:
Thu, 27 Jul 2006 14:46:02 -0300
Subject: Re: [asterisk-users] SIP Woes

> 
> Thanks for the suggestion, I added that in and now get:
> 
> 
> 	Jul 23 16:57:31 WARNING[4114]: pbx.c:1292 pbx_extension_helper:
> 	No application 'Dial' for extension (Outgoing, 100000, 1)
> 	Reliably Transmitting (no NAT):
> 	SIP/2.0 403 Forbidden
> 	Via: SIP/2.0/UDP
> 192.168.1.11:5064;branch=z9hG4bK7a6c25f1-041c-db11-82b2-000fea3f84d4
> 
> And, to make sure I didn't make a type in my dialplan:
> 
>         exten           =>      _X.,1,Dial(SIP/${EXTEN}@Sipgate,30,trg)
>         exten           =>      _X.,2,Hangup
> 
> 
> Any thoughts ?

app_dial.so is not loaded, so the Dial dialplan application does not exist. You can load it from the CLI by doing load app_dial.so or explicitly putting it in your /etc/asterisk/modules.conf to be loaded when Asterisk starts.

> 
> Dave
> 	

Joshua Colp
Digium