Hello,
I have MeetMe rooms generated dynamically and it always have two people
inside that are entered by dialplan.
I wish to make in some way a timeout mechanism that after X amount of time,
it will disconnect the users and kick them out of the conference.
How can I do such thing ?
Thanks,
Thank you all for the comments.
Jared, I've implemented your idea, and it worked very well. Thank you very
much for it :)
Ido
On Mon, Jun 1, 2009 at 10:36 PM, Jared Smith jsm...@digium.com wrote:
On Mon, 2009-06-01 at 10:22 -0500, Danny Nicholas wrote:
Write an AGI to hangup the users
Hello,
I have the following settings for manager on two Asterisk 1.2.24 (that
have installed over a year ago):
[user]
secret = password
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
write = call,command
On one server, Asterisk only react as you would expect - sending a
command without
On Sat, Apr 24, 2010 at 21:19, Christian christia...@runbox.com wrote:
Hi all,
Is anyone here using Asterisk on Archlinux?
Yes and no, I do use it on Archlinux for testing purpose but not as a
server.
Arch linux is not built to be a server distro, unlike Debian that have extra
steps for
Hello,
I'm trying to figure out what was the return code of SIP for a call.
The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to
retrieve the peer name using ${CHANNEL(peername)}, I have an error message
that CHANNEL does not have peername or it is not available to be used.
(SIP_CAUSE,${CDR(dstchannel)})})
Works fine on the Asterisk server I'm running (1.8.3.3).
Thanks, that works for me as well.
Philippe
Ido
On Mon, Jul 11, 2011 at 11:01 PM, ik ido...@gmail.com wrote:
Hello,
I'm trying to figure out what was the return code of SIP for a call
I'm using it.
Can you please provide more information on the issue with this feature ?
Is there another way to know the response code of SIP ?
Thanks,
Ido
On Thu, Aug 18, 2011 at 15:42, Matthew Nicholson mnichol...@digium.comwrote:
Greetings,
Recently a performance regression in chan_sip
Hello List,
I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works with FAX, Asterisk does
not detect any incoming fax. Even when I use 'fax set debug', it does not
display anything.
It's Asterisk 6.2.x . Any ideas what can I do to
] *On Behalf Of *ik
*Sent:* Thursday, November 17, 2011 5:30 AM
*To:* Asterisk Users Mailing List
*Subject:* [asterisk-users] Fax not detected by Asterisk
** **
Hello List,
I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works
On Fri, Nov 18, 2011 at 04:01, Edwin Lam edwin@officegeneral.comwrote:
On 11/17/11 3:30 AM, ik wrote:
I have an Elastix 2 machine with digium fax modules (with license).
When I try to create an extension that also works with FAX, Asterisk does
not
detect any incoming fax. Even when I
Hello,
I have a weird case, when some numbers dialed using a PRI, have an early
media sounds instead of normal ringing.
Few of the numbers are making Asterisk 1.6 (using Elastix 2) to report all
circuits are busy now. All of this numbers are cellular phones, but they
constantly reporting the same
Hello,
I have weird issue with Asterisk 8 lately.
When I call MixMonitor without mixing the channels, it changes the
sides of in and out.
Sometimes the first leg of the call is in and sometimes it's out.
I can't figure out if it's a known issue, or a new bug.
I'm using Asterisk 8.11.1
Any
, ik ido...@gmail.com wrote:
Hello,
I have weird issue with Asterisk 8 lately.
When I call MixMonitor without mixing the channels, it changes the
sides of in and out.
Sometimes the first leg of the call is in and sometimes it's out.
I can't figure out if it's a known issue, or a new bug
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