Thank for the hint. I will have a look into it.
Daniel
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von isr...@gmail.com
Gesendet: Freitag, 6. Mai 2011 15:22
An: Asterisk Users Mailing List - Non-Commer
Will this work:
exten=> 123,1,Meetme(1234)
exten=> 123,n,Hangup()
exten=> 5000,1,Dial(Local/123@bk_music/n,,m())
exten=> 5000,2,Goto(bk_music,123,1)
Parties can call 123 to enter a meeting room. and with the help of a
callfile ic an dial a local channel to 5000 extension which in return calls
a
Thanks for the reply. I looked into the G option of Dial applications. No
problem with that but How do I create a ghost call?
My dial plan will look like this:
Caller A calls Caller B normally:
exten=> _XXX,1,SomePreDialApps()
exten=> _XXX,n,Dial(SIP/B)
exten=> _XXX,n,Hangup()
Caller A calls ca
Run more of your systems as diskless. Make your tftp setup
indispensable :)
On Sun, 2011-05-08 at 22:37 +0100, Sebastian Arcus wrote:
> Hi James,
>
> Thanks for the reply. I'm not concerned about performance. But I've
> learned that every extra daemon software on a server comes with its
> secu
Hi James,
Thanks for the reply. I'm not concerned about performance. But I've
learned that every extra daemon software on a server comes with its
security caveats. I would feel much better about not having another one
to worry about and keep an eye on.
Sebastian
On 05/08/2011 10:30 PM, Jam
I have my tftp daemon running all the time and it really doesnt affect
the performance of the machine. Is there a reason why you want to shut
it down?
“I see blindness more as the ability and sight
more as the disability, I only see that which
is within a person.”
Patrick Henry Hughes - 2009
Hi all,
Sorry for posting here - but I figured there are many people with Cisco
IP phones here - and I use them with Asterisk :-)
I have a couple of Cisco 7940 phones. I've loaded the SIP firmware OK,
loaded the SIP configuration files OK, they work with Asterisk just fine.
My question is -
On Sat, 2011-05-07 at 16:24 +0100, --[ UxBoD ]-- wrote:
> I know a lot has changed over the past couple of years, and even
> monthly, and that Asterisk running within a virtualised environment is
> very happy indeed. If one would only be using SIP/IAX would Xen/KVM be
> the best solution ? / or per
On Sat, May 07, 2011 at 04:24:08PM +0100, --[ UxBoD ]-- wrote:
> I know a lot has changed over the past couple of years, and even monthly, and
> that Asterisk running within a virtualised environment is very happy indeed.
> If one would only be using SIP/IAX would Xen/KVM be the best solution ? /
Hi All,
I have set following milliwatt and dial in from my mobile phone. and i adjust
txgain value in chan_dahdi.conf but no effect on dahdi_monitor value is same.
even i change its to 20 and more but monitor still showing same value. But i
can hear tone variation on my mobile when i am chang
https://issues.asterisk.org/view.php?id=18868
-Original Message-
From: satish patel
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 8 May 2011 11:43:41
To: asterisk-users
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] no ri
Hi,
I have PRI configured and up but when i am dialing outside i am not getting any
ringback tone but my call is connected. following is my example
SIP->PRI > mobile
I have set progress=yes in chan_dahdi.conf but still not working
if i call inbound from my mo
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