asterisk users - Jan 2006 - Incoming PSTN Calls

Hi all,
 
I am having difficulty getting incoming PSTN calls working. I have set
up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
 
My provider told me to change my sip.conf as follows
 
register => username:password@sip.blueface.ie/2093                  

; To receive incoming calls specify this block and replace
"yourcontext"
for your dial plan. 
[blueface-in] 
type=peer 
host=sip.blueface.ie 
context=incomingpstn
 
And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)
 
[incomingpstn]
exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1)                    //press 1
for internal extensions.
 
 
This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and pointing it
to the appropriate context but that didn't work either. The only way I
can get it to work is to copy the code I had in the 'incomingpstn'
context of my extension.conf to the 'default' context. i.e.
 
[default]
exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1)        
 
Then the file would play. First of all I don't get why this is.It
doesn't even seem to refer to the code in my sip.conf.I don't get it.
Secondly whilst moving this code to the default context means I can hear
my initial welcome menu, when I press '1' to interrupt the menu and move
to menu option 1 (another sound file) it won't let me interrupt and I
eventually get the error "Timeout but no rule 't' in context
'default".
 
Does anyone have any ides where the problem might be?
 
Many thanks,
Aisling.


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Is Exist "InternalExtension" context ? and 2093 exten ?

2006/1/5, Aisling <ashling.odriscoll@cit.ie>:
>
>  Hi all,
>
>
>
> I am having difficulty getting incoming PSTN calls working. I have set up
> an account with a third party provider. In my system, the user register
with
> SER and use Asterisk for PSTN access, voicemail etc
>
>
>
> My provider told me to change my sip.conf as follows
>
>
>
> register => username:password@sip.blueface.ie/2093
>
> ; To receive incoming calls specify this block and replace
"yourcontext"
> for your dial plan.
> [blueface-in]
> type=peer
> host=sip.blueface.ie
> context=incomingpstn
>
>
>
> And then in my extensions.conf to have something similar to the following
> (or however I wanted to handle my incoming calls)
>
>
>
> [incomingpstn]
>
> exten => 2093,1,Wait(1)
>
> exten => 2093,n,Background(MainMenu)
>
> exten => 1,1,Goto(InternalExtension,2093,1)                    //press 1
> for internal extensions.
>
>
>
>
>
> This didn't work and I kept getting a 404 not found error saying the
user
> didn't exist. I tried creating the user in sip.conf and pointing it to
the
> appropriate context but that didn't work either. The only way I can get
it
> to work is to copy the code I had in the 'incomingpstn' context of
my
> extension.conf to the 'default' context. i.e.
>
>
>
> [default]
>
> exten => 2093,1,Wait(1)
>
> exten => 2093,n,Background(MainMenu)
>
> exten => 1,1,Goto(InternalExtension,2093,1)
>
>
>
> Then the file would play. First of all I don't get why this is?It
doesn't
> even seem to refer to the code in my sip.conf?I don't get it. Secondly
> whilst moving this code to the default context means I can hear my initial
> welcome menu, when I press '1' to interrupt the menu and move to
menu option
> 1 (another sound file) it won't let me interrupt and I eventually get
the
> error "Timeout but no rule 't' in context 'default".
>
>
>
> Does anyone have any ides where the problem might be?
>
>
>
> Many thanks,
>
> Aisling.
>  -------------------Legal
> Disclaimer--------------------------------------- The above electronic mail
> transmission is confidential and intended only for the person to whom it is
> addressed. Its contents may be protected by legal and/or professional
> privilege. Should it be received by you in error please contact the sender
> at the above quoted email address. Any unauthorised form of reproduction of
> this message is strictly prohibited. The Institute does not guarantee the
> security of any information electronically transmitted and is not liable if
> the information contained in this communication is not a proper and
complete
> record of the message as transmitted by the sender nor for any delay in its
> receipt.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
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>
>
>

--
Giovanni Miano
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