[Asterisk-Users] Incoming PSTN calls not routing to Asterisk? (using Sipura 3000)

Sharon Lim limleechin2005 at gmail.com
Fri Jun 16 19:32:45 MST 2006


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 at 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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