[asterisk-users] "Dropping call because extensions '200', 's' and 'i' doesn't exists"

[s m]

thanks guys, i solve my problem.

as Asghar said, i remove 2 and forget to add it again therefore
asterisk can not recognize extension 200 in extension.conf file.

this is my extension that works properly:
exten=>_2.,1,Dial(SIP/to-231/1${EXTEN:2})

thanks every body for your attention.
Sam

On 4/13/13, Gertjan Baarda <gertjan.baarda at gmail.com> wrote:
> Can you post both extensions.conf from both systems?
>
> Sent from my iPhone
>
> On 11 apr. 2013, at 14:51, s m <sam.gh1986 at gmail.com> wrote:
>
>> this is my [from-trunk] extension:
>>
>> [from-trunk]
>> exten=>_2.,1,Dial(SIP/to-232/2${EXTEN:1})
>>
>> and this is [to-231] in sip_additional.conf:
>>
>> [to-232]
>> host=192.168.0.232
>> type=peer
>> qualify=yes
>>
>> and 192.168.0.232 in the ip address of my freepbx.
>>
>>
>> On 4/11/13, A J Stiles <asterisk_list at earthshod.co.uk> wrote:
>>> On Thursday 11 April 2013, s m wrote:
>>>> when i call 100 from 200, every thing is ok and phone is ringing but
>>>> when i call 200 from 100, it says "service unavailable".
>>>>
>>>> i debug asterisk in my system 2 and see below message:
>>>> "Dropping call because extensions '200', 's' and 'i' doesn't exists
>>>> in context [from-trunk]"
>>>
>>> OK.  What do you have in the [from-trunk] context in your extensions.conf
>>> ?
>>>
>>>
>>> --
>>> AJS
>>>
>>> Answers come *after* questions.
>>>
>>> --
>>> _____________________________________________________________________
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>>
>> --
>> _____________________________________________________________________
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>
> --
> _____________________________________________________________________
> -- 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
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