[Asterisk-Users] Incoming PSTN Calls

[Aisling]

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.


-------------------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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060105/cff9c3ee/attachment.htm