We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in
April 2011 - basically now. After that, only security patches would be
committed. This is already a delay from the original plan published
Tilghman,
Could you remove your Reply-By header, please? Your deadline is two months in
the past (and in any case, list postings really shouldn't have a reply deadline
at all)
Here is your Reply-By header from your March 21 email:
Reply-By: Wed, 19 Jan 2011 16:20:00 -0600
Thanks!
Here is how I would do it:
First, come up with a numbering scheme. For instance, all extensions in
location 1 are 1xxx, extensions in location 2 are 2xxx. External calls are
9xxx-xxx-
In Location 1:
Sip Trunk 1 - goes to Location 2. The dial plan uses it as outgoing trunk for
all calls
Do you mean, using the smart phone as an Asterisk server, or as a device (i.e.,
an extension)?
I think running Asterisk in server mode would run up against blocking of SIP
traffic on most voice networks. Also, you would probably run into issues with
battery life, and with availability (what if
Use IPTables to lock down your machine to only accept incoming connections from
your local network and from the particular IPs that you are expecting
connections from (such as your SIP trunk, maybe).
That is of course assuming that these calls are made by SIP.
Don't forget to also change all
Basically, any door control system that works with DTMF tones should work - in
theory. You will probably need to play around with the length of the DTMF
tones, and maybe also with the level.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
There are many ways to do this, and very little information to go on.
For instance, if you have Exchange 2007 and a lot of money, you can integrate
it with Asterisk using Exchange UM (Unified Messaging). Microsoft charges extra
for the premium CALs you need to actually do that. There are also
I assume that you checked and the remote IP is a legitimate IP phone? If not,
it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration is correct
- if the SIP authentication fails, you can see this happening.
From:
I've had to rip out VoIP in two cable-modem situations because the call quality
was too poor.
Bandwidth isn't the main characteristic you are looking for; most Internet
connections have plenty of that. Latency and jitter matter far more. Latency
describes how long each packet travels from your
DSL to T1
On Sun, 2010-09-12 at 15:32 -0700, Kevin Keane wrote:
In terms of telephony, a T-1 can make a huge difference over DSL. DSL
gives you a lot of raw bandwidth, true, but for voice that really
doesn't matter all that much. Voice calls only take a relatively small
amount of bandwidth
-13 at 00:32 -0700, Kevin Keane wrote:
Latency also is the reason VoIP does not work at all over satellite
connections even though they tend to have plenty of bandwidth.
Please define does not work at all over satellite ???
Sure, it is not studio HIFI quality, but is th same quality as you
In terms of telephony, a T-1 can make a huge difference over DSL. DSL gives you
a lot of raw bandwidth, true, but for voice that really doesn't matter all that
much. Voice calls only take a relatively small amount of bandwidth anyway; you
can fit dozens of concurrent calls into a DSL or T-1.
to T1
On Sun, 12 Sep 2010, Kevin Keane wrote:
What really matters is the latency, and T-1 is a huge improvement over
DSL in that area. The easiest way to measure latency is the ping time
to a server that is “close to you” Internet-wise. A DSL has latencies
of between 40ms (if it’s
Are you talking about VMware Server, ESX/ESXi, or one of their other products?
The only VMWare product that I can even conceive might work is ESX/ESXi.
Others have already pointed out that in VMware, you won't get direct access to
the hardware. VMWare does have some limited capability to
Do you have a Nagios server? Then you could use that to monitor various aspects
of Asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Sunday, August 08, 2010 11:28 AM
To: Asterisk Users Mailing List -
From: John Novack [mailto:jnov...@stromberg-carlson.org]
Sent: Sunday, July 25, 2010 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Kevin Keane
Subject: Re: [asterisk-users] Using Vertical IP2007 phones with Asterisk?
Kevin Keane wrote:
I recently inherited a Vertical
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Monday, July 26, 2010 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fail2ban -
I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4
and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware.
The Asterisk setup relies heavily on queues with dynamic agents. The problem I
am having is that on SOME (but not all) the Cisco phones,
leave a queue
Check your dialplan.xml file that the affected phones are loading.
Thanks,
--Warren Selby
On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com
wrote:
I recently inherited an Asterisk system (PBX in a Flash, based on
Asterisk 1.4 and FreePBX). The phones are mostly
the same tftp address?
Thanks,
--Warren Selby
On Jul 25, 2010, at 4:47 PM, Kevin Keane subscript...@kkeane.com
wrote:
Stupid question (sorry, I'm pretty much an Asterisk beginner) - where
do I find the dialplan.xml? As far as I can tell, there is no TFTP
server in this network. I found
I recently inherited a Vertical Xcelerator IP system with IP2007 phones. I
would like to use the phones with an Asterisk system instead, but there doesn't
seem to be much information on it on Google. Is it even possible? These phones
claim that they are SIP phones.
Thanks!
Kevin
--
, Jul 25, 2010 at 9:52 AM, Kevin Keane
subscript...@kkeane.commailto:subscript...@kkeane.com wrote:
I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4
and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware.
The Asterisk setup relies heavily on queues
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