On Sun, Sep 25, 2011 at 09:39:06PM -0700, Steve Edwards wrote:
On Sun, 25 Sep 2011, Alex Balashov wrote:
Aside from that, is it really that big of a deal? Is it that hard
to learn a new command set and adapt?
Yes, it is.
I confess I'm a 1.2 Luddite so I have close to no experience with
On 02/10/11 21:58, dotnetdub wrote:
On 2 October 2011 21:36, Sebastian Arcus s...@open-t.co.uk
mailto:s...@open-t.co.uk wrote:
Just a follow up. I've opened up udp ports 1-2 on the Linux
box (where Asterisk is) and now I have sound. However, bear in mind
that the Netgear
On Sat, Sep 24, 2011 at 9:35 PM, Bruce B bruceb...@gmail.com wrote:
Hi everyone,
I don't mean to be rude but honestly which genius comes up with changing the
Word to the wise -- if one starts a sentence with I don't mean to
be...X your true intentions are to be just that.
If you find yourself
Greetings-
I'm working on a unique Asterisk installation where I've been given a
requirement of keeping a voice call active, even during a data connectivity
loss. So, let's assume I have remote users connecting to an Asterisk server via
sometimes unreliable connectivity such as satellite,
Assuming that you don't have some sort of reconnect protocol going on like
SIP headers, a native-bridge to a local channel might do the trick for you.
If you are using DAHDI, you might be out of luck.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Maybe you could use a very simple sollution like a meetme room - you have
only to be creative with the dialplan.
Ioan
www.modulo.ro
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On 11-10-02 07:53 AM, Sassy Natan wrote:
Hi Group
I have added the following to my /etc/apt/sources.list
deb http://packages.asterisk.org/deb natty main
deb-src http://packages.asterisk.org/deb natty main
deb http://packages.asterisk.org/deb natty-proposed main
deb-src
Hi All,
Trying to upgrade some call servers, in the lab making sure all my
applications work, ran into an issue with some manager perl scripts
that pull and reset database info, it seems the command and result
responses have changed but I'm not sure how to resolve. My scripts
are using CPAN
Hi
I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
I configured incoming calls from pstn to ring my SIP phone (extension : 100)
cat extensions.conf
...
[from-pstn]
exten = s,1,Dial(SIP/100,10)
same = n,VoiceMail(100,u)
root@PC-debian:/etc/asterisk# cat dahdi-channels.conf
...
...
