The AstLinux Team has released 1.2.0. All current users are encouraged to
upgrade as this release addresses the bash ShellShock bug.
New in 1.2.0:
* New Linux Kernel 3.2.x
* igb ethernet driver for Intel Atom C2000
* Enable AES-NI support
* New sip-user-agent firewall plugin
* New versions of
The AstLinux Team is happy to announce the release of 1.0.4.
New in this release:
-- Asterisk 1.4.44 and 1.8.14.1
-- DAHDI, dahdi-linux 2.6.1 and dahdi-tools 2.6.1
-- wanpipe, version bump to 3.5.27
-- rhino, version bump to 0.99.6b2. Support is now enabled again by default.
-- libPRI,
The AstLinux team is happy to announce the release of version 1.0.2. This
release features several security updates. All current users are encouraged to
upgrade as soon as possible. Please see the documentation at
http://doc.astlinux.org for upgrade details or the official release pages.
The AstLinux Team would like to announce the release of 1.0.1. This version is
available with either Asterisk 1.4.43 or Asterisk 1.8.8.3. A full changelog
and upgrade (or new install) instructions are available on our website. Please
follow the upgrade instructions carefully when upgrading
The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This
release includes significant changes and improvements over past releases.
Specific upgrade and new installation instructions are available at:
http://www.astlinux.org
Some of the highlights include:
* Using eglibc
The AstLinux Team would like to announce the immediate availability of
the 0.7.8 release. This release includes either Asterisk 1.4.41 or
Asterisk 1.8.4. All current users are encouraged to upgrade to this
release to take advantage of bug fixes and other updates to Asterisk.
Please note that
be able to adapt it
for your use, but as it's written, it's integrated with AstLinux.
http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/0.7/package/arnofw/adaptive-ban/
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
a remote unit that
connects to the network and a local device at the door, giving some
better security.
I've used the Valcom VIP-172 phones. They are simple and work well.
Very good support if you need to call them.
http://www.valcom.com/Home_links/sipdoorintercom.htm
Darrick
--
Darrick
The AstLinux Team would like to announce the immediate availability of
the 0.7.7 release. This release includes either Asterisk 1.4.40 or
Asterisk 1.8.3. All current users are encouraged to upgrade to this
release to take advantage of bug fixes and other updates to Asterisk.
PPTP was added
The AstLinux Team is happy to announce the latest release (0.7.6).
There are several security updates as well as feature
enhancements/improvements. All current users are encouraged to update.
A full changelog is available at:
http://www.astlinux.org
Both Asterisk 1.4.39.1 and Asterisk
The AstLinux Team is happy to announce the release of AstLinux 0.7.5
with options for both Asterisk 1.8.1.1 and Asterisk 1.4.36. More
information about the release is available on our website:
http://www.astlinux.org/content/astlinux-075-release
Direct links to the installation files are
many bugs and security issues that have been addressed in
newer versions. 1.4.37 is the latest version from the 1.4 branch. It's
quite possible that whatever you're trying to fix is already fixed in
that newer release.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
The AstLinux Team is happy to announce the release of AstLinux 0.7.4.
This is a dual release which allows you to chose between Asterisk 1.4.36
or 1.8.0.
There are several security updates and other improvements. All current
AstLinux users should upgrade as soon as feasible.
One of the more
and can be found for around $90.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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not changed since months.
Any help is appreciated
Best regards,
Patrick
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New
.
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New to Asterisk? Join us for a live introductory webinar every Thurs
=6087294351
dtmfmode=inband
dtmf=inband
canreinvite=no
[guest]
type=friend
host=dynamic
canreinvite=no
context=DID_trunk_1
--
Darrick Hartman
DJH Solutions, LLC
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The AstLinux Team is happy to announce the release of AstLinux 0.7.3.
This update contains mostly bug fixes and security updates. All current
users of AstLinux are encouraged to update to this release.
Updating can be performed from the web interface or from the command
line using a few
-implementations like MasterShaper, if
it works well together with a firewall that uses iptables.
Kind regards,
Jonas.
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Darrick Hartman
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because there is a long instance of silence sometime.
Thanks
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after X amount of time, disputing false claims etc). Maybe someone
could contact spamhaus to create a list for VOIP since they seem to have
a nice system in place?
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
On 04/12/2010 12:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project. It's a
blacklist of IP addresses and address ranges which are known to ONLY be
used for malicious purposes
kicked the
bucket. He's shuffled off his mortal coil, run down the curtain, and
joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing
priority)
mediagw*CLI
Asterisk: 1.6.2.6
--
Darrick Hartman
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Sometimes you need to look at the cost to pull new wire too, not just the cost
of the phones. There are a few cases where the channel banks + analog phones
make sense, especially when the analog devices are already there.
Sent from my BlackBerry® wireless device from U.S. Cellular
to check two more items:
Upgrade firewall plugins:
upgrade-arno-firewall check
upgrade-arno-firewall upgrade
Install sound files:
**NOTE sound files are not installed by default starting with 0.7.1**
upgrade-asterisk-sounds upgrade core en ulaw
Enjoy!
Darrick
--
Darrick Hartman
DJH Solutions, LLC
AstLinux is well under that. You could build a custom image that
contains only what you want and have it under 30M. We have support for
sqlite3, but not mysql or postgresql. You would have to build your own
package to include python. Our build environment is based on buildroot,
but has
The AstLinux Team would like to announce that the 0.7.0 version of
AstLinux is available for download. There have been many significant
updates in this release including updating to the latest Asterisk
Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other
system updates.
For a
a pattern which
matches 6662020 while you do have something that matches 0216672020.
Without seeing the dialplan, we can only guess.
--
Darrick Hartman
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-purpose
RJ-45 jacks to work with POTS lines. Run everything down to a central
panel and send pots over the jacks that you need to. That way if you
decide you need/want to go IP in the future, you're all set.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
not in a properly cooled
environment. 1U and 2U servers work well in data centers with 60F
cooling. Not so much in your normal office with computers crammed in a
closet.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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of postage but I really think SIP
is your silver bullet.
That would be nice of you, but he should find out the problem before
throwing more hardware at the issue.
Darrick
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
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actually pretty easy using unionfs.
Darrick
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit
On 12/20/2009 11:38 AM, meetmecall wrote:
I used msmtp for delivering mail and this is the procedure I documented
once, based on info I found on the internet. I hope it is of help.
msmtp also has a rudimentary 'queue' option if you use the msmtpQ/msmtpq
scripts
--
Darrick Hartman
DJH
On 12/02/2009 08:32 AM, Martin Roy wrote:
I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b
option that let you enter the first name OR last name of a user. I see that
to make this work I need a patch. I'm wondering how can I install this patch
as it's an option one
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been working with Asterisk for some time.
However, I don't seem to be able to put a call on
Mike Diehl wrote:
On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote:
Mike Diehl wrote:
On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote:
Mike Diehl wrote:
On Friday 27 November 2009 11:09:02 am Noah Miller wrote:
Hi Mike -
I've got a Polycom 501 that's been
Also install a recent version! 1.4.26.3 would be the latest in the 1.4
release series. Using something as old as 1.4.13 is not recommended.
Alex Balashov wrote:
You need to install 'gcc' and 'g++' and associated libraries and headers.
hadi motamedi wrote:
Dear All
Please be informed
No problems resolving anything here. Website appears to be up. Maybe
they had a temporary equipment issue.
Matt Darnell wrote:
On Wed, Nov 11, 2009 at 1:11 PM, Matt Florell astma...@gmail.com wrote:
They had a nice booth at Astricon and everything. Haven't heard anything
about them going
John Timms wrote:
Hi. I'm having trouble figuring out why I'm not able to make many
concurrent VoIP calls on my system. I'm not aiming for a huge number,
because I have purposely bought a low powered system, but I would
think that I could get more. Here are the details:
I have a
Ott Rose wrote:
I have question thats not really about astrisk but I figure you guys are
doing this sort of thing.
We use Aastra 6757i phones. there is some support for XML. the question
is how would i go about learning to customize these phones?
Read the manual on the Aastra website.
Russell Horn wrote:
Hi,
I've a DID number that gets passed to three internal phones and a cell
phone via my outbound IAX trunk. If the cell phone is off or out of
coverage, its voice mail captures the call.
What's the best way to avoid this? Is there a recommended way to force
the cell
aster...@opensourcesolution.in wrote:
hello all,
friends i am new in asterisk. i had just finished the installation
requirment of asterisk. i am using Centos 5.3 in which ill be installing
asterisk now guys plz guide me my requirment for deploying asterisk is,
i am having a
Oliver,
I have some comments, but am looking for a good answer to this as well.
1). 2.6.27 kernels to include a version of mISDN v2.0. It's not as up
to date as the full version on the mISDN.org site.
2). From my understanding chan_lcr is the preferred way to connect
Asterisk to mISDN
Barry L. Kline wrote:
Kevin P. Fleming wrote:
It's not present in the current 1.4 doc/imapstorage.txt file, or any
later version. I don't even know why the storage format would matter,
since that would be very specific to the IMAP server that is managing
that folder.
Hmmm
John Todd wrote:
On Oct 17, 2009, at 7:47 PM, Michael Graves wrote:
I'm told that they will show up on the event site in about three
weeks.
On Sun, 18 Oct 2009 02:29:48 + (UTC), Jeff LaCoursiere wrote:
Wish I could have made it :( Is there a possibility of a
collection of
the
Randy R wrote:
On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
| I have three Snom M3s at the moment but getting pretty fed up with
| the
| issues :( I am UK based and would be interested to hear of other
| peoples
The S685IP has no headset jack AFAIK. If you
John Knight wrote:
You know, I'm not entirely sure. I've never thought about using it
outside the context of Tomato. Does anyone else know if that's a
standalone (and hopefully architecture independent) package?
Michelle Dupuis wrote:
I like the Qos functionality. Is that a linux based
das sandesh wrote:
Hi All,
Today I faced a problem with channels getting stuck. We use asterisk
1.4.18.1, and there were 2 extensions (channels) that got stuck. When I
try to do soft hangup channel, it says Requested for soft hangup
for that channel, but if we go and check once again
Add this line to your aastra.cfg file
sip intercom allow barge in: 0 # don't barge in on existing calls
Olivier wrote:
Hi,
When implementing click2dial feature, I can trigger an Aastra phone to
auto-answer using statement like :
SIPAddHeader(Alert-Info: info=alert-autoanswer);
This
Use the 'hidefromdir' option in voicemail.conf for that particular
entry. This should be clearly documented in the example voicemail.conf
file.
Michelle Dupuis wrote:
I have internal mailboxes that I don't want visible to callers going
through the directory. Is it possible (in * 1.4) to
Ira wrote:
I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no
problems and while everything seems fine i get these message at
startup and than all is well. Should I be worried or do i need to let
the team know about this?
Also, is not finding /dev/dahdi/transcode a problem
Polycom sip.cfg is not the same as the Asterisk sip.conf file
hadi motamedi wrote:
Thank you for your reply . Please find attached my Asterisk sip.conf .
Can you please let me know what modifications are needed ?
Regards
H.Motamedi
On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney)
Does anyone have a working patch for the following issue on Asterisk
1.4.26 or an earlier version of 1.6 than 1.6.2? It looks like it got
committed somewhere after 1.6.1 was branched and is only available
natively in Asterisk 1.6.2.x.
https://issues.asterisk.org/view.php?id=12382
I have a
The Astlinux Development Team is happy to announce the release of
AstLinux 0.6.7. This release is a security and bugfix release with no
new features. All current users of AstLinux are encouraged to upgrade.
Current users can upgrade either from the web interface or by issuing
the following
I still don't see what you gain by using m0n0wall and a separate
Asterisk install. I can't think of one thing that you would need a
separate m0n0wall instance to do that AstLinux can't do on it's own.
The web interface has become quite completely in the last few releases.
Traffic shaping,
Brian McEntire wrote:
Darrick -
You seem adamant, and I will look deeper into the firewall in Astlinux! :-)
Brian,
I am one of the developers, so I happen to like what we've done. There
have been some huge changes to the web interface and the overall project
in the past year or so.
Alex Samad wrote:
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
Alex,
Here's a good place to start.
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conversation. Let me know if your tests are successfull.
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...there is something in the way the RTP packets are sent/received by
Asterisk and maybe it can be correlated to the missing audio.
Giorgio
Darrick Hartman (lists) wrote:
Do you have 'canreinvite=no' in your sip.conf entry for this phone? If
not, you should.
On 06/18/2009 07:55 AM, Giorgio
Jeff,
Contact their tech support. You will need to send the card in for
service, but they may be able to repair it. You should look into
getting some sort of surge protection on the analog lines if you don't
already have something. The surgegate stuff seems to work well.
Darrick
On
There were some serious issues with some of the earlier 1.4.x Asterisk
releases. You say it's a production server and can't upgrade because of
that. That is the one reason why you should upgrade. There are
security risks with certain versions and some serious bugs that were
fixed. While I
On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote:
On Thu, 4 Jun 2009, Rob Hillis wrote:
Jeff LaCoursiere wrote:
We are still talking about a $175 phone. How about the Polycom IP 320?
$85 at 888voipstore. Can't go wrong with Polycom for voice quality.
True, Polycom's are brilliant for voice
Do you have any idea the number of bugs that have been fixed since
1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this.
On 06/02/2009 08:58 AM, Adrian Marsh wrote:
Hi,
It's a 2mb dedicated leased fibre line, with50% utilisation.
My first thoughts were the internet
The AstLinux project is happy to announce the latest release of
AstLinux. Astlinux-0.6.6 is now available for download or upgrade.
New users should go to the http://www.astlinux.org website and follow
the instructions for a new installation.
Current users can upgrade their existing 0.6.x
Asterisk 1.4.25 does work with Zaptel.
On 05/31/2009 07:46 PM, bilal ghayyad wrote:
Hi All;
I discovered that Asterisk 1.4.25 does no thave zapata.conf, any advise? Does
it mean that Asterisk 1.4.25 no more support for zaptel and it works only
with dahdi? So, what is the latest Asterisk
On 05/21/2009 09:11 AM, Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Robin Rodriguez wrote:
still rather frustrating getting the EFK working. If needed I could
post that portion of sip.cfg to get you started.
Please do! Just having the example could be helpful for
Gordon Henderson wrote:
On Wed, 6 May 2009, Alan Lord (News) wrote:
On 06/05/09 08:28, Gordon Henderson wrote:
snip /
One little tip: You need to compile Asterisk for an i586 processor as
the VIA processor is missing a few (mmx, etc.) instructions that a full
blown i686 has.
Hi Gordon,
it. It may have been discussed during a VUC session or may
have been on this list.
Either way, I'm unable to google my way to it. Can anyone point me in
the right direction?
That would be Karl Fife, of the famous Karl Fife experience.
http://kfife.com/voip/
Darrick
--
Darrick Hartman
DJH
Rob Hillis wrote:
Kurian Thayil wrote:
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
Daily Asterisk restart
Do you think its mandatory in production env?
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups
Rob Hillis wrote:
Darrick Hartman wrote:
Rob Hillis wrote:
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation spikes have
decreased significantly.
Unless you're using some unstable modules, there really
The latest version of Astlinux has been released today.
Changes include:
Asterisk 1.4.24.1 (security fix AST-2009-03)
Asterisk-gui svn 4739
Astlinux web interface 1.4.07
Update Zoneinfo to '2009e'
Additional checking is now done when performing an upgrade of the OS.
Existing users of Astlinux
Gabriel - IP Guys wrote:
Dear All,
I have a asterisk setup that is currently running on version 1.4.15 – I
wish to upgrade or migrate this instance to the current asterisk stable,
1.6.0.6. It is my intention to build a FC8 box, then install asterisk
from source, and begin to
Mike wrote:
I'm looking for a good network device that does bandwidth management.
It can be integrated in a router or stand-alone, but must be SIP-friendly.
I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest
hardware revisions can't downgrade to the version that
Roland Roland wrote:
Hi all,
a few month ago I got the task of setting up asterisk for my company.
I had 94 employee to set this up for ...
I never heard of asterisk before to b honest, so after researching a bit..
I started with a digium card with ZAP
though that didn’t work out as the
The openfire project has this functionality as part of their package.
Requires a Tomcat install, but it works. I set it up on my website as
an example, but haven't used it much. (It does work nicely though).
Don't see what this has to do with Asterisk though.
Darrick
Dean Collins wrote:
The AstLinux Team is happy to announce that AstLinux 0.6.4 is available.
All users of AstLinux are encouraged to upgrade since this release
fixes the recently reported security vulnerability in Asterisk 1.4.23.1
Right now a mix up on the Sourceforge site is preventing us from
uploading full
Leif Madsen wrote:
Darrick Hartman wrote:
I know the call parking feature changed in 1.4.23.1 to fix some serious
issues. I'm seeing a major change though which I find disturbing.
A person parks a call by transferring it to the parking position (700).
When the timeout value is reached
I know the call parking feature changed in 1.4.23.1 to fix some serious
issues. I'm seeing a major change though which I find disturbing.
A person parks a call by transferring it to the parking position (700).
When the timeout value is reached, the call is NOT returned to that
device, but
On another topic, I would say those gateway are not so easy to configure :
- a web server is embeded but it is not documented anywhere and it's
GUI is far from natural,
- alternatively, you can edit a config file for which a huge doc is
available but, as this boxes are not specifically
We are proud to release Astlinux 0.6.3. All users of AstLinux should
upgrade to this release. Files are available for download at the
Astlinux SourceForge project page.
https://sourceforge.net/project/showfiles.php?group_id=170462
Updates include new versions of Asterisk, Asterisk-gui,
Doug wrote:
At 12:33 2/22/2009, michel freiha wrote:
Hi all,
I took my decision to use Asterisk server for handling my VOIP
calls...My next step is to choose the best hardware that I should
use i order to have the best performance...Here I faced 2 choices
for my hardware (CPU)...
1-
Jeff LaCoursiere wrote:
On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote:
Hello Asterisk Users and those with an Interest in VoIP Tech,
[snip]
Is there a Chicago area users group? If not is there any interest in
creating one?
We have a group in Milwaukee that meets monthly
OCG Technical Support wrote:
A little off topic but
I need to put a 24 port Gig PoE switch into a small office – no computer
room / rack etc. All CAT5 terminates near the owners desk (smart huh?).
I want to put a PoE switch in place, with 24 ports and Gig speed.
Everyone
I have an interesting observation which I thought I'd pass along to save
other people from spending time trying to 'fix' it.
One of my clients uses Charter's so called business phone service.
They provide 'analog' phone lines over IP. In general, they've worked
OK. End users were saying that
On Tue, 30 Dec 2008 16:51:55 -0500, Doug Lytle supp...@drdos.info wrote:
That can't be correct.
Doug
Could be. It's been quiet all around with the holidays.
Darrick
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Hi Folks,
Had a quick search through the archives for softphones and cannot see any
recommended ones.
My question is what recommended free softphones are out there that can be
used with Asterix? I don't really know how many are out there. Is anyone
currently using a softphone with
Actually, it could be within Asterisk, but only if you have Zaptel
hardware. If you are only using SIP devices, then the problem is with
the phone configuration. You really don't provide enough information to
determine what is causing your problem. How are you provisioning the
phones? What
Philip Prindeville wrote:
Just saw from build 2036:
Now, to get the following packages to build:
misdn
asterisk-chanmisdn
nistnet
rhino
strace
rp-pppoe
Whoops. I'm sure Philip thought he was sending this to a different
mailing list.
Time for you to discover who's your dahdi...
Asterisk 1.6 used dahdi and not zaptel.
--Original Message--
From: Christian
Sender: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
David Gibbons wrote:
Two separate networks? Did I miss something? I feel like I'm taking
crazy pills! Two separate physical networks means twice the hassle,
twice the maintenance, twice the cost, twice the headache. Not to
mention the fact that the whole idea of VOIP is to simplify IT and
If you buy your phone from a reputable place they will be able to provide the
firmware.
--Original Message--
From: Andrew Joakimsen
Sender:
To: Asterisk Users Mailing List - Non-Commercial Discussion
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Vieri wrote:
--- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote:
Why not swap it all with just IP phone?
That's because we have almost 400 analog phones already wired in our building
(which is very large). So we need to take advantage of the wiring.
Also, if we were to convert to an
Olivier wrote:
Hi,
Whenever I'm logging in with asterisk -r command, I can see that the
verbosity and debug levels are set to a value which is different from
the last ones I left when I logged off from CLI.
Where are those default levels defined ?
I can't see any related option in
...since everyone else top posted.
Take a look at the application setmusiconhold.
CLI core show application SetMusicOnHold
You can use this in a dialplan as follows:
[tenant1incoming]
exten = s,1,Wait(1)
exten = s,n,Answer()
exten = s,n,Background(tenant1sounds/welcome)
exten =
Dean Collins wrote:
I'd also like to know what happens when someone 'chats' to the account
connected to the Asterisk server.
Lots of questions about this one. There's definitely a demand for it so
I can see why Digium would be interested in exploring this option. Time
will tell how well it
looking for a solution that can be used to monitor Asterisk and
the
Telco lines aswell as the network (Servers, WAN LAN links, Router
Switches)
We use nagios for that.
--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com
http://www.djhsolutions.com/wiki
Michael Graves wrote:
On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
Asterisk should work fine with any phone that supports that codec.
Personally, I have only used it with Polycom phones. Also, again,
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has
Jay R. Ashworth wrote:
On Wed, Aug 13, 2008 at 11:01:46PM -0500, Darrick Hartman wrote:
You can get an adapter for the Plantronics that will plug into the 2.5mm
jack on the phone.
I need the opposite adapter: to plug a 2.5 headset into an RJ-9
Polycom.
Anyone know where I can find
Matt Riddell wrote:
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Paul Hales wrote:
That's a good question - the plantronics are available with
interchangeable ends - which makes them easy to move between different
phones.
Problem is, the headset button only works for the minijack
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