Ok, but thats static routing. My architecture is this:
[pbx extensions] --- [SIEMENS PBX] ---- [ASTERISK] --- [SER] --- [sip
clients]
I can't put in Asterisks sip.conf the hundreds of pbx extensions (and
they are always changing), I must do a dinamic forward for all 74XXX calls.
I think this is realy close:
exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
because it seems that is everything right... but It always answer:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
extension 's' in context 'default', but no invalid handler
Joao Pereira
Moises Silva wrote:
its kind of weird may be the problem is the default context, i have
never used the default context, i always use a specific context for
each extension. Lets say you have a registered sip number 21, then you
can do in sip.conf
[21]
someparameter=blah...
etc...
context=sipcontext
the important thing is the parameter called 'context' it has as value
'sipcontext'. When the extension 21 calls, then the dialed number (any
number the extension 21 dials) will arrive to the specified context
'sipcontext'. in sipcontext you write
[sipcontext]
exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
that should work. let us know if you still have problems.
On 7/29/05, Joao Pereira <[EMAIL PROTECTED]> wrote:
but everytime I dont put the "s", when I try to call 74XXX, Asterisk
answers :
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
extension 's' in context 'default', but no invalid handler
I think it must be something like that:
exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
... but it always answers:
pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid
extension 's' in context 'default', but no invalid handler
It must be a way to do it...
Thanks
João
Moises Silva wrote:
Please read this docs:
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
you need to understand what the 's' extension does. If you use it, no
matter what number they have dialed, it will start at the s extensión.
If i understand your goal, YOU DONT NEED the 'exten => s,1,Answer' .
You have:
;exten => s,1,Answer
;exten => _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r)
please replace it for:
exten => _74XXX,1,Answer()
exten => _74XXX,2,Dial(SIP/[EMAIL PROTECTED],30,r)
best regards
On 7/29/05, Joao Pereira <[EMAIL PROTECTED]> wrote:
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:[EMAIL PROTECTED]
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:[EMAIL PROTECTED] ... not
yet what I want
;exten => s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
; this way, he dials "sip:[EMAIL PROTECTED]" ...
;exten => s,1,Dial(SIP/[EMAIL PROTECTED],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/[EMAIL PROTECTED],30,r)
; what must I put here to dial sip:[EMAIL PROTECTED] ???
-------------------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 <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] ????
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/[EMAIL PROTECTED],30,r)
but this way all calls go to [EMAIL PROTECTED] .....
Then I tried:
exten => s,1,Dial(SIP/[EMAIL PROTECTED],30,r)
but this way, the system tries to dial <sip:[EMAIL PROTECTED]> and not
[EMAIL PROTECTED] 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
[email protected]
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
[email protected]
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
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users