[asterisk-users] SIP Woes

[Dave Hope]

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Joshua Colp wrote:
> ----- 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.
> 

Wow. How did I miss that one :)

Thanks for your help!


>> Dave
>> 	
> 
> Joshua Colp
> Digium
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