I installed Asterisk and a digium wildcard (X100P). Using
the extensions.conf with a few changes and a sip.conf file
that includes two extensions, I can place calls between the
SIP phones. I also can call in to the SIP phones from the
PSTN using the X100P. On incoming calls I can hear the
default d
show dialplan will show the asterisk view of the dialplan.
show channels will display channels in use and
sip debug will show what the sip phones are doing.
Also, have a console open as this often provides clues, especially if
started with some verboseness -vv
You may try making a more ge
thanks for responding.
the changed the include commands and they are now at least
causing the extension to match using one of the local
10-digit numbers. this is what shows up on the console:
Executing StripMSD("SIP/1008-32df", "1") in new stack
-- Executing Dial("SIP/1008-32df", "Zap/1|BYEXTENSIO
Just FYI stop using BYEXTENSION because it will be going away soon.
use ${EXTEN} or ${EXTEN:x}
bkw
- Original Message -
From: "Tom Scott" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 01, 2004 12:29 PM
Subject: Re: [Asterisk-Users] dialing out t
rom: "Tom Scott" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 01, 2004 12:29 PM
Subject: Re: [Asterisk-Users] dialing out to PSTN from SIP phones
thanks for responding.
the changed the include commands and they are now at least
causing the extension to match
On Saturday 01 May 2004 09:42 pm, Tom Scott wrote:
> okay, will use ${EXTEN}.
>
> it all seems to be working now. I think my problem was
> understanding the flow of control using contexts, but
> i also needed to do some reading on syntax and variables
> -- and more to come.
>
> the working commands
