[Asterisk-Users] Incoming PSTN Calls - Stumped

[Iqbal]

I had a similar problem , and then used GoTo instead of include

Iqbal

Aisling O'Driscoll wrote:

>Hi,
>
>Yes InternalExtension is the context and 2093 the extension.
>
>Just to explain something odd that’s happening (and I’m very stumped
>with this)….I think my contexts are definately the reason that I
>can’t interrupt the menu for incoming pstn calls to choose a submenu:
>
>My users register with my sip proxy (SER). Therefore when I create an
>entry for them in sip.conf I set only one context. Also to allow for
>incoming calls from my provider it seems I must direct the calls
>firstly to a ‘dummy’ extension.
>
>sip.conf
>
>register => username:password at provider.ie/2093
>
>[provider-in]
>type=peer
>host=sip.provider.ie
>context=onecontext
>
>[2092]
>type=peer
>other stuff
>context=onecontext
>
>So the dummy extension here is ‘2093’ and 2092 is a phone who
>registers with SER and when SER redirects to Asterisk uses the
>‘onecontext’ context.
>
>Now in my extensions.conf ‘onecontext’ includes other contexts. This
>is how I get access to conference calls, creating IVR menus etc. Also
>the main purpose of ‘onecontext’ is to allow outgoing access to the
>PSTN.
>
>[onecontext]
>include => createmenu		//creating an IVR menu
>include => createconf		//creating a conf call
>etc
>include => default		//used for voicemail
>
>[createmenu]
>;does something
>
>[createconf]
>;does something
>
>;outgoing calls – main purpose of onecontext
>exten => _X.,1,Dial(SIP/${EXTEN}@provider-out)
>exten => _X.,2,Hangup
>
>[default]
>
>;mailbox for 2092 and other users
>
>
>Now this is where the problems start! For incoming calls I tried to
>do “include => incomingpstn” in ‘onecontext’ which I thought would
>call a new context called ‘incomingpstn’ which would have an entry
>for the dummy user. i.e.
>
>[incomingpstn]
>
>exten => 2093,1,Wait(1)
>exten => 2093,n,Background(MainMenu)
>exten => 1,1,Goto(InternalExtension,2093,1)    //directs to another
>context called Internal Extension
>
>I also changed the [provider-in] for context=incomingpstn in my
>sip.conf. However this didn’t work and I kept getting directed to the
>voicemail of my pstn provider. The ONLY way I could get the incoming
>calls working was to add the contents of the ‘incomingpstn’ context
>to the default context i.e.
>
>[default]
>
>exten => 2093,1,Wait(1)
>exten => 2093,n,Background(MainMenu)
>exten => 1,1,Goto(InternalExtension,2093,1)    //directs to another
>context called Internal Extension
>
>With this I can hear the MainMenu when I dial my DDI but I can’t seem
>to interrupt to divert to another submenu. In the testing that I have
>done the user that is making the call is 2092 registered with SER. If
>I change the context of 2092 directly in sip.conf to incomingpstn,
>then I can hear the menu and interrupt to go to the submenu. But
>obviously then I don’t have access to the other features in Asterisk.
>The point is that I’m stumped as to why it only works in the default
>context and if this is the case how do I get it to call the submenu.
>
>This is what comes up on my asterisk console:
>-- Executing Dial (“SIP/2092-2829”, “SIP/021123456 at provider-out”) in
>new stack
>-- Called 021123456 at provider-out
>-- Playing ‘MainMenu’ (language ‘en’)
>-- other messages (not relevant I think)
>== Spawn extension (outgoing, 021123456, 1) exited non-zero on
>‘SIP/2092-5837’
>== Spawn extension (default, 2093, 2) exited non zero etc etc
>
>I’m very stuck on this and can’t figure it out.
>Any help appreciated.
>
>Many thanks,
>Aisling.
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>Giovanni Miano
>Sent: 05 January 2006 21:09
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Incoming PSTN Calls
>
>Is Exist "InternalExtension" context ? and 2093 exten ?
>2006/1/5, Aisling < ashling.odriscoll at 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 at 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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