[Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main Menu

[Aisling]

Hi Kokmeng,

Unfortunately that's wasn't it. WaitExten was executed but then I still
get the timeout error - 

Timeout, but no rule 't' in context 'incomingpstn'

I am totally stuck...I have been googling and searching the archives and
testing different things for days to no avail. I thought at some stage
it might be an issue with the priorities and all different priorities
but that didn't work either. 

I see the Asterisk console play the MainMenu (i.e. the Background rule),
I press an option and absolutely nothing appears on the console, the
menu carries on regardless. Its only at the end I see this timeout
error.

Thanks,
Aisling.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of KokMeng
Loh
Sent: 11 January 2006 01:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Can't interrupt Main
Menu

Hi Aisling,

You're missing the 'WaitExten' directive after playing the background 
sound file. Your lines should be like this:

[incomingpstn]
exten => s,1,Wait(1)
exten => s,n,Background(MainMenu)
exten => s,n,WaitExten(10)
exten => 1,1,Goto(internalExt,s,1)
exten => 2,1,Goto(mainconfmenu,s,1)


-kokmeng.

Aisling wrote:

>Hi,
>
>Thanks to both Iqbal and Kokmeng for the replies. 
>
>Kokmeng I tried what you suggested however no luck...
>
>What I have done which is currently working(kind of) is that in my
>sip.conf in the [general] section I have set context=incomingpstn. My
>register line looks like:
>
>register => username:password at sip.provider.ie/
>
>In my extensions.conf I then have
>
>[incomingpstn]
>exten => s,1,Wait(1)
>exten => s,n,Background(MainMenu)
>exten => 1,1,Goto(internalExt,s,1)
>exten => 2,1,Goto(mainconfmenu,s,1)
>
>[internalExt]
>exten => s,n,Background(InternalExtension)
>
>[mainconfmenu]
>exten => s,n,Background(MainConfMenu)
>
>I can hear the MainMenu sound file being played. What's strange is that
>when I press '1' to interrupt, which in my logic should invoke the
>internalExt context, nothing happens. The MainMenu sound file continues
>to play and finally I get the error:
>
>Warning: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context
>'incomingpstn'
>
>I used the 'Goto' as Iqbal suggested instead of includes...
>
>Has anyone ever experienced this kind of behaviour before?
>
>Many thanks,
>Aisling.
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of KokMeng
>Loh
>Sent: 09 January 2006 08:53
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Incoming PSTN Calls - Stumped
>
>Hi,
>
>The hostname that you used in your register directive ('provider.ie') 
>must have a corresponding section in sip.conf. In your example, you
used
>
>'[provider-in]'. If that is what you actually used, then this might 
>explain why your incoming goes to the default context because it 
>couldn't find its own "section". Try renaming '[provider-in]' to 
>'[provider.ie]'.
>
>-kokmeng.
>
>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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