[Asterisk-Users] dialplan defenition
Christian Victor
christian at victormedia.de
Thu Jul 28 07:11:04 MST 2005
So just don't send them to extension s but extension _74XXX
Christian
Joao Pereira schrieb:
> I had tried that also, but it didnt work. In that case, if I dial 74118
> (for example) Asterisk answers this:
>
> pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/0' sent into invalid
> extension 's' in context 'default', but no invalid handler
>
> I think it needs the "s"... but how do I put the "s" and route the call
> to 74XXX at 193.136.252.5 ????
> Thanks
> Joao
>
>
> Christian Victor wrote:
>
>> Joao Pereira schrieb:
>>
>>> Im writing my dial plan, in witch every SIP phone begins with 74 and
>>> has more 3 numbers (like 74XXX).
>>> So, I want to route all 74XXX calls to my sip channel. For this I
>>> wrote this line:
>>> exten => s,1,Dial(SIP/74118 at 193.136.252.5,30,r)
>>>
>>> but this way all calls go to 74118 at 193.136.252.5 .....
>>>
>>> Then I tried:
>>> exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r)
>>>
>>> but this way, the system tries to dial <sip:s at 193.136.252.5> and not
>>> 74XXX at 193.136.252.5 like I wanted...
>>
>>
>>
>> You were on the right way my friend. Why not try
>>
>> exten => _74XXX,1,Dial(SIP/$(EXTEN)@193.136.252.5,30,r)
>>
>> Hope that helps
>> Christian
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
More information about the asterisk-users
mailing list