[asterisk-users] Incoming calls not being answered by asterisk
Roberto Milani
roberto.milani at sbcglobal.net
Sat May 24 09:56:05 CDT 2008
Ciao Roand
I think you should buy a book and do some reading to build up your
knowledge.
but in the meantime try something like this in the dialplan
(extensions.conf)
exten => PSTN,1,Answer() ; Answer inbound calls or internal miss-dials
exten => PSTN,2,Playback(silence/1)
exten => PSTN,3,Background(enter-ext-of-person) ; input an extension
exten => PSTN,n,WaitExten(20) ; Adjust wait, default 5 sec
exten => PSTN,n,Goto(internal,${EXTEN},1) ; Goto the correct extension
exten => PSTN,n,Hangup() ; End the call
where PSTN is your sipura SIP name (1002 i think)
Ciao
Roberto
On May 24, 2008, at 3:09 AM, RoLaNd RoLaNd wrote:
> Hello all,
>
> ive got the following setup currently:
>
>
> __Sipura 3102-----PSTN
> |
> Lan |
> |
> |__asterisk
>
> i configured both asterisk and pstn to be able to receive/make calls
> through each other using sip of course..
> but the thing is i want asterisk that when it receives an incoming
> call from sipura, to answer it, play msg that i recorded and wait
> for the caller to dial in an extension, where it would transfer the
> caller to that exntension, and in case the extension owner isnt
> available to answer it would direct him to his voicemail(tht i dont
> know how to set yet), and in case the caller didnt dial any
> extension in a certain amount of time, it automaticly directs it to
> a specific extensions i'd specify..
>
> i tried the examples given in lots of forums and so on none of them
> worked, the phone keeps on ringing with every incomign dial plan ive
> specified without asterisk answering it..
> the thing i did is that sipura directs incoming calls to 1002, so
> ive set the context of 1002 in sip.conf to a dial plan of [incoming-
> sipura] and ive set the commands i mentioned earlier tht i took out
> of those forums.. but theyre not working!!!
>
> anyone has an example i could go on with ?
> any help would be apreciated:)
>
> Discover the new Windows Vista Learn more!
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080524/537c65f1/attachment.htm
More information about the asterisk-users
mailing list