[Asterisk-Users] dialplan defenition
Joao Pereira
joao.pereira at fccn.pt
Fri Jul 29 02:41:18 MST 2005
Ok, now ill explain my dialplan problem
Goal: When Asterisk receives a 74XXX number, should send it to its peer
in 193.136.252.5:5060 (SERs IP), someting like:
sip:74XXX at 193.136.252.5
Here is my extensions.conf and sip.conf
------------------- EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp
TRUNK=CAPI
[default]
; this way he works... but always dials sip:74118 at 193.136.252.5 ... not
yet what I want
;exten => s,1,Dial(SIP/74118 at 193.136.252.5,30,r)
; this way, he dials "sip:s at 193.136.252.5" ...
;exten => s,1,Dial(SIP/${EXTEN}@193.136.252.5,30,r)
;this way it works... but I have to dial:
; 74XXX then he gives me dialtone, and then I must dial 74XXX again...
; not yet what I want... the idea is just dial 74XXX once, withou
dialtones in between
;exten => s,1,Answer
;exten => _74XXX,1,Dial(SIP/${EXTEN}@1193.136.252.5,30,r)
; what must I put here to dial sip:74XXX at 193.136.252.5 ???
-------------------SIP.CONF
[general]
context=default
port=1720
bindaddr=193.136.252.5
insecure=very
realm=fccn.pt
;defenition of SER as a peer
[193.136.252.5]
type=peer
username=193.136.252.5:5060
host=193.136.252.5
context=from-sip
canreinvite=no
insecure=very
Thanks
Joao Pereira
-----------------------------------------------------------------------------
Moises Silva wrote:
>the problem is how are you getting there? i mean, what do you have in
>sip.conf and please post all the relevant text in extensions.conf, not
>just the 'exten => blah' part, we need to know context names to see if
>its matching the sip.conf configuration
>
>regards
>
>On 7/28/05, Joao Pereira <joao.pereira at fccn.pt> wrote:
>
>
>>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