In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally it
is all TDM so there is absolutely zero latency.
We are able to use ATA's for faxes and analog phones but devices that use
modems, they fail 99.99% of
Aloha,
We are looking to roll a solution that will have the following network layout:
ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax
Does version 1.8 with the Digium fax driver have this capability? I
like 1.8 because it is a long term support version.
What ATA's are people using?
Any working
On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
da...@klaverstyn.com.au wrote:
I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and
rx_fax on multiple installations with no problems.
David,
Are you running 10.0 or 1.8?
Glad to know that the PAP2T has a solid T.38
When i reload asterisk, calendar show calendars does not show this.
What I am missing? I really need to get this to work!
You are missing that you should take out passwords from config files.
Hope your gmail account didn't get hacked.
--
On Sat, Mar 5, 2011 at 8:54 PM, Pezhman Lali l...@lopl.net wrote:
Dear
this note is only for fresh administrators don't think about asterisk
security.
Do you know where you go to 'un-ban' an IP if they made some mistake?
Using webmin I was not able to find the IP address that was was banned.
On Mon, Mar 7, 2011 at 9:15 AM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
iptables -L -v
will give you the IP address that was banned
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Aloha,
We have added the ability to dynamically forward or send a voicemail
to more than one mailbox.
Here is the link -
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18835
There is a diff file and a drop-in replacement for app_voicemail.c.
Here are some basic instructions for
Aloha,
Is there a way to forward a message to multiple people from within the
telephone user interface? Now there is only the ability to forward to
an individual.
I see there is a way to leave a message for multiple people using the
dial plan but that is not available when you are listening to
On Mon, Nov 15, 2010 at 1:04 PM, Matt Darnell mattdarn...@gmail.com wrote:
Is this command the best way to access a MySQL database -
MYSQL(Connect connid dhhost dbuser dbpass dbname) ?
I thought I heard that using ODBC was a bit more stable.
Anyone have any experience?
Thanks,
Matt
Is this command the best way to access a MySQL database -
MYSQL(Connect connid dhhost dbuser dbpass dbname) ?
I thought I heard that using ODBC was a bit more stable.
Anyone have any experience?
Thanks,
Matt
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Steve,
Did you use this syntax to convert:
sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql
-Matt
On Sun, Oct 31, 2010 at 9:43 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music
We have a customer that does not care for the default MoH.
We have downloaded some royalty free music but it sounds 'fuzzy' when we
test it with the system.
We down sample it to 16bit, 8KHz, Mono. We have tried with Audacity,
CoolEdit Pro, VLC.
Does someone have a file they can send me that
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.comwrote:
On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but it sounds 'fuzzy' when we
test it with the system.
Can you post a link to the original?
Here is the original - http
You'll also need to make sure you're properly reporting device state to
asterisk. I think this means you need to set a call-limit for each sip peer
that you want to monitor in sip.conf (we use 25 so there are no accidental
limits actually applied), and setup hints in your extensions.conf
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 10-10-15 04:10 AM, Сикорский Сергей wrote:
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any alternative for obsolete call-limit option in
We have a queue that agents log into through the dial plan. Extension
Sip/101 logs in as Agent/101
We have 'ringinuse = no' in the queues.conf file.
The issue is that when Ext 101 is on a 'non queue' call (they placed a
call, someone called their DID, etc) they still receive queue calls.
Is
.
Thanks,
--Warren Selby
On Oct 14, 2010, at 10:13 PM, Matt Darnell mattdarn...@gmail.com wrote:
We have a queue that agents log into through the dial plan. Extension
Sip/101 logs in as Agent/101
We have 'ringinuse = no' in the queues.conf file.
The issue is that when Ext 101
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote:
What version of asterisk are you using and method are you using to login your
agents? I recently had this issue with a 1.4.33 install where the agents
logged in with agentcallbacklogin. In the end I had to move them
On Fri, Oct 8, 2010 at 5:16 AM, Sebastien Thomas li...@amplisys.ca wrote:
One more thing: Make sure that the port going to your data-DHCP server
doesn't have the voice VLAN set on it. I troubleshot an installation for a
few hours before thinking of this...
Interesting, the DHCP server for
On Wed, Jun 30, 2010 at 12:10 PM, CunningPike cunningp...@gmail.com wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
Thank you Andrew,
I will check it out. We are currently running 1.4.
-Matt
On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham lath...@gmail.com
On Wed, Jun 30, 2010 at 4:26 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
Thank you Andrew,
I will check it out. We are currently running 1.4
OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Mon, Jun 28, 2010 at 8:40 PM, Matt Darnell mattdarn...@gmail.com wrote:
Is is possible with a Polycom phone to update
Is is possible with a Polycom phone to update the LCD with the
callee's name after dialing them?
When you dial ext 103 now, it says 'To:103'...would be nice if could
have 'To:Dan Marino'
This is the case even when you have a contact for ext 103.
None of the phones I have ever tested do this,
On Fri, May 7, 2010 at 3:41 AM, Jared Smith jsm...@digium.com wrote:
To make it more clear and less cryptic, we split out the callcounter
functionality in sip.conf, so that you could turn on/off the SIP device
state tracking without limiting calls, and encouraged people to use the
GROUP() and
Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
-Matt
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Most manufacturers charge in excess of $80 to upgrade from a 10/100
switch to a 10/100/1000 switch built into the phone.
The cost might have been in the chipset 5 years ago but I can get a 5
port gigabit switch for $30.
What are most folks using for people that need gigabit to the desktop
and
On Thu, Jan 21, 2010 at 3:30 PM, Jonathan Thurman jthurma...@gmail.com wrote:
On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell mattdarn...@gmail.com wrote:
Most manufacturers charge in excess of $80 to upgrade from a 10/100
switch to a 10/100/1000 switch built into the phone.
The cost might have
On Wed, Dec 30, 2009 at 8:11 AM, Thermal Wetland
thermalwetl...@gmail.com wrote:
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP
Anyone know what happened to netxusa?
Seemed like they dropped off the web overnight.
-Matt
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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 down, this might just be an unfortunate IT management
incident.
Both their toll free and fax numbers go to a re-order
A client of mine asked about a Polycom IP321..anyone else heard about it?
-Matt
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I wish Polycom would hire someone with ergonomics skills. The whole
menu system is the most painful ever designed outside entry-level
phones. Polycom is an acknowledged leader in sound quality and robust
hardware but their idea of a menu sucks rocks and always has. Most of
their menus require
Yes with EFK in the latest firmwares you are able to change the on
screen button layout. I used it to bring a Do Not Disturb button to
the main screen of the SoundPoint IP330's. I may just be dense but
paired with the Administrator and Developer guides from Polycom it was
still rather
Has anyone been able to do the following:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
2. Put Record on the main screen when a call is active. This would
eliminate having to press the 'more' softkey.
Thanks,
Matt
So, in short, if all my calls were from outside to a G729 enabled phone and
vice versa, I would reach the limit at 30/30, NOT 15/15.
If you had 30 licenses, yes the limit would be when you needed either
30 decoders or 30 encoders. i.e. 1/30 would max you out.
-M+
On Wed, Jul 16, 2008 at 3:07 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Tue, 15 Jul 2008, Matt Darnell wrote:
Does anyone know of a bandwidth test that tests the upload with the download?
All of the ones I can find will test the upload then the download.
I from experience I have found
Does anyone know of a bandwidth test that tests the upload with the download?
All of the ones I can find will test the upload then the download.
I from experience I have found that a 3M/768K DSL can only do about
256K/256K simultaneously.
The only way I have of testing it is with FTP uploads
IP670 was just released...about 30% more than the IP650.
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip670.html
-Matt
On Tue, Apr 29, 2008 at 1:02 AM, Patrick
[EMAIL PROTECTED] wrote:
On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote:
Anyone seen anything
On Tue, Apr 29, 2008 at 1:02 AM, Patrick
[EMAIL PROTECTED] wrote:
On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote:
Anyone seen anything on the IP670 the Color Expansion?
Great timing. Yesterday I was looking at the IP650 and wondered when the
successor to the IP650 would arrive
Anyone seen anything on the IP670 the Color Expansion?
-Matt
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Any know what Digium hasn't released the DS3 card?
It was supposed to be out a while ago.
-Matt
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On Feb 1, 2008 3:53 PM, Thermal Wetland [EMAIL PROTECTED] wrote:
Hello,
On our Polycom phones we can not activate the Buddy Watch feature.
When you add or edit a contact, the list ends at Auto Divert.I know
it is the end of the list b/c the down arrow on the right side of the screen
On Nov 28, 2007 8:48 AM, Mindaugas Kezys [EMAIL PROTECTED] wrote:
Pros:
1. No need to reload Asterisk when you change settings
Is reloading the text based config that dangerous? Is there a memory leak
or something?
How many times can you reload before you should restart Asterisk?
-Thermal
Hello,
I have two Asterisk boxes, each in a different office. Extensions are 1xx
2xx in office 1 and 3xx if office 2.
I have setup IAX2 trunks between them as well as the Outbound Routes.
Intra-office dialing works great.
I can figure out how to transfer an incoming SIP call to the other
Matt,Wouldn't they have to actually enter a forwarded number for the forward to activate? I've hit the forward button myself many times after a call ends, and the phone asks you for a new number to forward to.
Douglas.You are correct you have to enter something as the contact and then press
Aloha,Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to.The real issue is the 'end call' button becomes 'forward' when the call endstherefore the user thinks they are pressing 'end call'
http://www.bethephonecompany.com/documents/itexpo_la/DSC00495.JPGYou need to have been around in telephony for a little while.
Aloha,Matt
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:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Matt Darnell
Sent: Wednesday, November 02, 2005
9:53 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Anyone
know who is in this picture?
http://www.bethephonecompany.com/documents/itexpo_la
I'll second that. Make sure your script is in
/var/lib/asterisk/agi-bin and you have the right permissions on it. I
really just wanted to reply to your post though to congraduate you,
Dan Marino, on your recent induction into the Pro Football Hall of
Fame ;)
Sorry, wrong Dan Marino!
-Dan
On 8/10/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Dan Marino wrote:
I have installed the Perl library from
http://asterisk.gnuinter.net/asterisk-perl and am wondering how I
reference agi-test.agi from extensions.conf
I have added
exten = s,1,AGI,agi-test.agi
but that doesn't seem
http://sites.gizoogle.com/?url=http://www.voip-info.org
-Matt
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From the Wiki:
'There are aready a few bugs in 1.5.2 but more fixes some good new
features'
Anyone know where the bugs are being listed?
I am working through a few issues:
1. When rebooting, the phone will pause for exactly 180 seconds with
the screen reading 'updating initial
From the Wiki:
'There are aready a few bugs in 1.5.2 but more fixes some good new features'
Anyone know where the bugs are being listed?
I am working through a few issues:
1. When rebooting, the phone will pause for exactly 180 seconds with
the screen reading 'updating initial configuration'. I
These phones are mentioned in the Sip 1.5 manuals, anyone know what
the differences are?
Where are you getting SIP 1.5 from?
When I log into the Polycom download area, all I can find is 1.4.1.
They must have pulled it back.maybe some issues, like 1.3.0
-Matt
These phones are mentioned in the Sip 1.5 manuals, anyone know what
the differences are?
Aloha,
Matt
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On 4/19/05, Mike [EMAIL PROTECTED] wrote:
. close source and we own the code.
You are no better then Microsoft.
Speaking of an over reaction
-Matt
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Aloha,
Issue:
Someone calls into voicemail and hangs up
Asterisk does not get the disconnect signal
Asterisk records for 10 seconds then hangs up
Problem:
Asterisk will send the voicemail to the user
The email reads that the message is 10 seconds long
The email attachement is only about 300
Im facing a strange problem using a linksys-pap2 (two ports) ATA: I cant
have two simultaneous incoming calls when i use g729 codec, if i use g711
(alaw) there is no problem, is this a know issue or am i missing something?
The PAP2 only supports one G.729 call at a time.
Same as the Sipura
Hello,
Does anyone know of an * plug in that will prompt a user for an
account code when they make a long distance call?
I see where you can have a static variable, but I am looking for a
lawyer bill back type application.
Thanks,
Matt
___
Does anyone know of an * plug in that will prompt a user for
an account code when they make a long distance call?
Look at the Authenticate command.
Do you know if the entered string gets printed out with CDR records?
Got this from the Wiki
1. accountcode: What account number to
On Mon, 28 Mar 2005 14:29:15 -0600, Jerry [EMAIL PROTECTED] wrote:
On Mar 27, 2005, at 12:10 AM, Matt Darnell wrote:
Has anyone been succesful pushing a VLAN setting to a Polycom phone
via DHCP?
Chicken or the egg! How can the Polycom reach the proper DHCP server
Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP?
I can push the boot server via option 66 but that is about it.
I have set it for 'fixed' and tried many different option numbers with
a couple differnet DHCP servers.
SIP firmware 1.3.4 or 1.4.1 doesn't make a
Has anyone been succesful pushing a VLAN setting to a Polycom phone via
DHCP?
Chicken or the egg! How can the Polycom reach the proper DHCP server
if it is not on the correct VLAN? That's why Ciscos and Polycoms
support CDP, so the CDP-capable switch can supply the correct voice VLAN.
There was some talk last June about some folks trying GR303 with *.
Asterisk supports GR-303 access concentrators now; I do not know if the
support is in stable, or only in CVS HEAD.
Asterisk does not know how to act _as_ an access concentrator, however.
Do you have an recomendations for
Aloha,
There was some talk last June about some folks trying GR303 with *.
Was anyone succesful?
Would love to hear about it.
-Matt
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Simple Answer: No. the i2004 uses the proprietary nortel UNISTIM
protocol. Asterisk uses SIP, IAX, SCCP, H.323, but not UNISTIM.
Complex answer: It depends on how much you really want it. There has
been an open-sourced implementation of a UNITSTIM server done by
Cedric Hans. It
Aloha,
Did anyone ever get the formating manual for the XML brwoser on the
Polycom IP600?
Does anyone have a sample?
Aloha,
Matt
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To
Did anyone ever get the formating manual for the XML brwoser on the
Polycom IP600?
Does anyone have a sample?
I'm using the Polycom micro browser big time... I have the parking, the sip
users online, agents on queue, the meetme rooms and the calls joined.
I don't have the manual... I call
I can't make anything appear in the screen.
I have tried tags like:
titlehello world/title
bodyhello world/body
texthello world/text
and on and onanything I can think ofjust produces a blank screen
Maybe I am missing a header or something. The Cisco XML was relativly
I am embarased to say that I changed it from 456. Can't seem to find
the paper it was written on! :(
Hi Matt,
Press and Hold: 4, 6, 8, * until it reboots.
Once you press this keys, you get a prompt for the admin password!
Once you press the 4,6,8, * you can enter the MAC
Does anyone know how to default the admin password on a Polycom IP500?
Phone has SIP load 1.3.1
Thanks,
Matt
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Does anyone know how to default the admin password on a Polycom IP500?
Phone has SIP load 1.3.1
I have physical access to the phone.
Thanks,
Matt
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On Fri, 10 Dec 2004 01:26:32 -0500, Brent Franks [EMAIL PROTECTED] wrote:
I think it is 456
- Brent
Does anyone know how to default the admin password on a Polycom IP500?
Phone has SIP load 1.3.1
Thanks,
Matt
Brent,
I am embarased to say that I changed it from 456. Can't seem
On Fri, 10 Dec 2004 01:36:22 -0500, Brent Franks [EMAIL PROTECTED] wrote:
Brent,
I am embarased to say that I changed it from 456. Can't seem to find
the paper it was written on! :(
-Matt
Hi Matt,
Sorry I read your last message too quickly. There is an admin guide at
On Tue, 7 Dec 2004 12:58:11 -0500, George Herndon
[EMAIL PROTECTED] wrote:
On Dec 7, 2004, at 8:48 AM, [EMAIL PROTECTED]
wrote:
ken ,
i too have a comdial analog pbx. i'm running a seperate vm system and
would like to migrate to asterisk. right now, my comdial
hands off calls via
On Thu, 02 Dec 2004 17:31:44 +1100, David Uzzell
[EMAIL PROTECTED] wrote:
Has the wiki died or is it just my routing to the wiki from Australia?
I have not been able to connect to it for the last hour or more :(
David
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Aloha,
I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN.
I have an Asterisk box, RC2 with a for port FXS card providing
dialtone for a Norstar Key System.
I have it working so when you press a line key on the Norstar you get
dial tone from the Asterisk box. The user has
Will you post to the list?
-Matt
On Fri, 20 Aug 2004 15:32:35 -0500, John Baker [EMAIL PROTECTED] wrote:
Still waiting on Polycom for something. Will make it available as soon
as I get it.
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I had this exact issue. The 7.1 firmware has some issue where it
won't upgrade a SCCP image. I had to upgrade to phone to SIP 6.3 then
to 7.1
That was the key for me.
-Matt
On Fri, 20 Aug 2004 17:49:42 -0400, Doug Shubert [EMAIL PROTECTED] wrote:
what version of SCCP are you running?
Cisco
That was part of my problem.
I can now get the 600 to download XML, I tried using
http://phone-xml.berbee.com/menu.xml and the phone displays XML Error
(1,0) syntax error. I'm guessing this is because the XML files at that
location are formatted for the Cisco phones. Anyone have
Is the conference going to be recorded for later playback.
They keynotes and conferences would be nice as well.
Aloha,
Matt
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Can anyone get to www.nufone.net?
Is their VoIP down?
-Matt
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PROTECTED] On Behalf Of
Matt Darnell
Sent: Sunday, 13 June 2004 5:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is nufone web site down?
Can anyone get to www.nufone.net?
Is their VoIP down?
-Matt
I don't know about their VoIP, but their site works for me.
-Shaun
Aloha,
Does anyone know how to have a hot keypad on a Cisco 7960?
It allows you to dial on-hook without press the SPEAKER button. Very handy
once you get used to it!
-Matt
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The in-between fix really wasn't a fix. Chan_sip was modified some time
ago
(5/24 or so) to require authentication for inbound calls also. To turn
this
required authentication off, you need to add insecure=very to your peer
definition.
5/24 is some time ago? The age we live in!
-M
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