asterisk users - Jun 2006 - Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)

Incoming calls from my Sipura 3000 don't seem to be correctly routing
to Asterisk (or something?)

Here is my Asterisk configuration for my incoming PSTN line:
Code:

[1000]
type=friend
host=dynamic
context=incoming
secret=6769
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very


Inside of extensions.conf, I have this:
Code:

[incoming]
exten => s,1,Answer( )
exten => s,2,Background(enter-ext-of-person)


When I call my PSTN line, my Sipura 3000 seems to successfully answer
it because the line rings once, but then immediately switches to a
second dial tone. Shouldn't my incoming call be answered and then have
"enter-ext-of-person" played to them?

What could be causing this?

Also, on a side note, I have a context called [home] which each SIP
Phone is associated with.  Do I need to specify each extension in
there?

For example:

exten => 50,1,Dial(SIP/50)
exten => 50,2,Hangup

exten => 21,1,Dial(SIP/21)
exten => 21,2,Hangup

Can't I just setup a default system where any two-digit number is
assumed to be an extension and it is automatically tried?

Thanks for any help!!

Hi John,

Your first question, I am not sure why ....but for this part i can explain
abit
> Also, on a side note, I have a context called [home] which each SIP
> Phone is associated with.  Do I need to specify each extension in
> there?
>
SIP user can register as name as well . Doesnt means to have number. Example
in sip.conf
[john]
type=friend
username=john
host=dynamic
context=incoming
secret=6769
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very

Then in extensions.conf , you can have any number to ring this john sip user
phone. Example :

exten =>9XXXXXXX,1,Dial(SIP/john) ; any number start with 9 end with 7 digit
behinds.  or you can also
exten => 9XXXXXXX,2,Hangup

exten => s,1,Dial(SIP/john) ; starting of the incoming call will ring John
phone.
exten => s,2,Hangup

Hope my explaination is clear or fullfill your needs....thanks

On 6/17/06, John Klimek <jklimek@gmail.com> wrote:
>
> Incoming calls from my Sipura 3000 don't seem to be correctly routing
> to Asterisk (or something?)
>
> Here is my Asterisk configuration for my incoming PSTN line:
> Code:
>
> [1000]
> type=friend
> host=dynamic
> context=incoming
> secret=6769
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> insecure=very
>
>
> Inside of extensions.conf, I have this:
> Code:
>
> [incoming]
> exten => s,1,Answer( )
> exten => s,2,Background(enter-ext-of-person)
>
>
> When I call my PSTN line, my Sipura 3000 seems to successfully answer
> it because the line rings once, but then immediately switches to a
> second dial tone. Shouldn't my incoming call be answered and then have
> "enter-ext-of-person" played to them?
>
> What could be causing this?
>
> Also, on a side note, I have a context called [home] which each SIP
> Phone is associated with.  Do I need to specify each extension in
> there?
>
> For example:
>
> exten => 50,1,Dial(SIP/50)
> exten => 50,2,Hangup
>
> exten => 21,1,Dial(SIP/21)
> exten => 21,2,Hangup
>
> Can't I just setup a default system where any two-digit number is
> assumed to be an extension and it is automatically tried?
>
> Thanks for any help!!
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