asterisk users - Nov 2006 - Call-limit

hello all,
 
isit possible to do sth. so that if the sip-phone is in use and a call
is incoming that the caller gets a busy signal?
coz i wan to have the incoming party get a busy signal if i'm at phone.
 
regards rene


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Hi for all!
	I have two questions about IAX2 connections ...

- is it possible to use codecs used by IAX2 depending on SIP codecs?

I have some SIP phones and one FAX using IAXMODEM. SIP users use alaw 
and IAXMODEM use slin. When i'm calling out thru IAX2 connection the 
best solution is to use slin codec for fax and GSM for SIP (with 
asterisk transcode), but i don't know how to configure this, because all 
connections going thru IAX2 use codecs based on IAX2 peer configuration 
(all calls go thru GSM)....

- is it possible to set TRUNK=yes if im using IAX2 realtime?

I can't find "trunk" field in iax_buddies table description.


thanks for all comments, cheers Marian


-- 
Marian Rychtecky
marian@mari.cz

Tel. +420 724 397 441
ICQ 76582857