You want to use Authenticate() between answer and dial.
http://www.google.com/search?q=asterisk+authenticateie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
Thanks,
Steve Totaro
On Sun, Aug 24, 2008 at 11:26 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hi all,
i;m obviously
Hello Roland,
You can use the cmd Read for this.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
Pretty straight forward. Whenever you need to accept DTMF input from the user
collect the required digits using Read; check the collected digits; if yes jump
to required extension; else
Hello Steve,
thanks for the advice :)
though one prob! if i add the authenticate line itll require all callers to
enter 1234 to access *ANY* sip account..
even though this would come in handy at some point but at the moment i just
want to deny the extension 300 from being able to call 01
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all
I have one solution in mind, maybe it is an overkill but:
You can create a db entry for each sip account, DB(family/key) lets name
family=destination sip number and key=${Callerid(num)} and assing a value 0
or 1, so string will be like this DB(301/300)=1 fot that 300 sip account,
and for all
Roland,
The simple solution is to utilize the power of contexts (put exten 300
in a different context in sip.conf or db) and includes to separate yet
include 300 (so 300 can be called and call other internal extensions).
Add authenticate before the dial statement.
The easiest way to do it, is
