Rushowr wrote:
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benjamin Jacob
Sent: Monday, September 04, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] includes in realtime ??
Hello ppl,
Is it possible to include contexts in the RealTime scenario??
If not, wots the work around??
Thanks in advance.
Ben.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Amazing how the wiki has this vast amount of AT LEAST info to start your
research on
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Sorry mate.
Just slipped the eye.
Now to another question, which I tried about.
With the Realtime arch, can we change parameters of certain users, say
sipusers, at runtime, for e.g. the codec and the change being reflected
back immediately?
The two SIP users I had, had allow set to "gsm;g729;ulaw;alaw", and the
two Xlite phones have gsm,ulaw and alaw configured.Calls work fine .
I changed the codec(set allow to have only g729). But still the calls
go thru.
I tried realtime load sipuser name <username>, to no effect. (anyway,
realtime load is only for reading values, if i am not wrong).
So is it possible to change user parameters at realtime?
or am I missing something again?
Thanks again.
Ben.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
