Does anyone know if there is a way to continue in the dial plan for the
called (outbound) channel if the caller channel disconnects? Something
like this:
*
[call_client]
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:1},30,g)
exten => _9NXX,2,Playback(some_fil
Looking for a configuration for the "Dial" application that is simular to
the "g" option, but for outbound, not inbound.
Is there a way to configure an outbound call sequence that will continue
to the next priority in the dialplan when the originator disconnects? The
senerio is an outbound call c
Is there a way to have a Meetme room dial an extension? For example, is
there a way to use the meetme as the channel in an originate command?
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I am havng the same issue even after "rm -r -f
/usr/src/asterisk-1.2.0-beta1" and tar extract and recompile. (make
clean;make install). This tarball was downloaded Tuesday afternoon.
Asterisk appears to be running normally, except show version "Asterisk
built by root@ on a x86_64 running Linux
Date: Tue, 30 Aug 2005 08:56:33 -0500
Asterisk Supporter wrote:
> Asterisk has this error on compile:
>
> flex ast_expr2.fl
> "ast_expr2.fl", line 50: unrecognized %option: reentrant
> "ast_expr2.fl", line 51: unrecognized %option: bison-bridge
> "a
Asterisk has this error on compile:
flex ast_expr2.fl
"ast_expr2.fl", line 50: unrecognized %option: reentrant
"ast_expr2.fl", line 51: unrecognized %option: bison-bridge
"ast_expr2.fl", line 52: unrecognized %option: bison-locations
make: *** [ast_expr2f.c] Error 1
2.6.12-1.1447_FC4smp #1 SMP
I am about to build a Dual Opteron Asterisk box as our soon to be
production server.
Is Core 4 supported or should I stay with Core 3?
There was a recent post about an issue with the latest Core 3 Kernel and
zaptel. I had the same experience, but just rolled back to the previous
version of the Ke
How "real time" is realtime? If the extensions.conf is stored in the
database, does * query it row by row or is it "cached"? In other words,
given the following exerpt:
exten => 5001,1,Dial(IAX2/[EMAIL PROTECTED]/s,30,g)
exten => 5001,2,Voicemail(u5001)
exten => 5001,102,Voicemail(b5001)
exten =
Anyone running Asterisk on dual Opteron Server? Are there any special
issues in a 64 bit environment and what is the capacity curve like?
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1) Trying to create a browser based "Click-to-Call" feature for * that
appears to the user as a hands free call on Cisco 7960 phones (SIP). If I
use the Action: Originate function, the phone does not auto answer, but
rather rings and if answered initiates the call. If I manaually change
the line
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