Hello Matthew,
It is always nice to see improvements. I look forward to testing your
patches.
It just seems that so many other hardware manufacturers have tackled the
problem, I am surprised digium has not put more research into getting
the issue solved in software, which is possible, as opposed
On Fri, Oct 28, 2005 at 10:19:54AM +0600, Tharanga wrote:
do i have another way to use this new TDM04B card with aterisk,zaptel. 1.0.7
version .
OR can i use 1.0.7 on one end and othe end 1.0.9 ?? is it make any
problem..to my IP Calls Via IAX2 channel .
You can upgrade zaptel alone. No need
Yes I do, but it has been installed automatically with asterisk, since I use
FreeBSD ;). Sending FAX works without problems from my SIP ATA's.
Bye, florian
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Andy Kuo
Gesendet: Freitag, 28.
Can anyone out there help me ?
Beside countless benefits, without Caller-ID on my
SIP Phone's screen * is a real pain in my neck.
I have * 1.2.0-Beta (Latest CVS) and a X100P card.
Sometimes I can get the Caller-ID and sometimes I can
not. Even from the same number. Any suggestions ?
On 10/27/05, Erick Baum [EMAIL PROTECTED] wrote:
We're having a rather serious echo problemusing the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it
Hello.
I have installed asterisk 1.0.9.dfsg-5 in debian sarge.
if I run /etc/init.d/asterisk stop and then /etc/init.d/asterisk start .
Asterisk don't detach from console where i started it. It beguin to write all
warning,debug,AGI dialog,... to the console.
If I start ast. manually without
Thanks everyone for the feedback on this. I'd say early next week
for an alpha release. We have decided to release it under the BSD
license, and it will go up on rubyforge after the first release.
Chris
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Erick, we're also using 1.0.1.12, having some echo problems,
mostly with in/out going ZAP calls (on quadBRI, w/asterisk
1.0.9), the internal SIP calls seem to work fine. (but you
have to make sure your volume isn't too high) Also the
GXP-2000 has the annoying feature that calls get
On Friday 28 October 2005 07:30, C F wrote:
As I said it could exist, but I'm only guessing here that the posts
about ADSI over SIP channels are (again this just my guess) only for
the SIP channel to allow for the ADSI scripts to be downloaded into
the phones. Since it's like faxing that
Just saw this thread.. Wanted to know if you'd like some input from me...
I'm developing ARTCP for controlling, managing, and end-user access to
Asterisk RealTime
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-astgroups
-Sent: Wednesday, October
Hello,
After further checking I found that when activating the firewall no traffic
is allowed OUT from the box.
Nameresolving, http, nothing accept ICMP works, even though I added:
iptables -A OUTPUT -p all -j ACCEPT
So I think its not related to asterisk at all, rather some iptables config
Hi
I have a Sipura SPA 2000 unit and I have configured both the lines in the
unit. both the lines are configured to use 729.
when I make calls from the lines independently it works great. no problem at
all.
when line 1 is connected and when I try to make a call using line 2 while
line 1 is
Eric Bishop wrote:
/etc/asterisk/
/usr/sbin/safe_asterisk
/usr/sbin/asterisk
/usr/lib/asterisk/modules/
/usr/include/asterisk/
/lib/modules/`uname -r`/misc
/usr/lib/
/usr/include/
Anything I have missed?
/var/lib/asterisk
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I have a strange problem with IAX.
We have an IAX connection between two boxes which works
fine. We can call from the two boxes, in and out using SIP phones, hard and
soft without issue.
We are testing some IAX softphones, and have come across a problem
with the voice. Calls on the
I'm trying to get my queue to exit and go to voicemail when the sip
phones that belongs to the queue is not registered (aka turned off).
Any suggestions?
I've tried setting leavewhenempty = yes. But it didn't work.
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Hello all,
I'm trying to find an Asterisk web interface (or windows gui interface) to
asterisk that won't allow users to go making changes to config files. I've
trawled through the very extensive list in the wiki, but there doesn't seem
to be a clear defining line between applications that are
Hi all, does anyone know if there is any app/webui that can show phones
that are currently registered to *. I guess this sort of funcionality
counld be grabbed from the CLI with iax2 show peers and sip show peers,
but having little programming knowledge wouldn't know where to start.
I'm
Couldn't find anything on the lists or in Wiki..
Customer wants to be able to dial complete SIP URL's... from his SNOM
phone.
ie - He dials on his phone [EMAIL PROTECTED] (which is more
difficult than a Number - but not undo-able)
How do I configure my extensions.conf to handle this sort of
Is there an estimate on how many calls a 2Mb ADSL line can handle at the same time?
Bearing in mind that the upload speed is 256Kb.
Thanks
Dan
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Asterisk-Users mailing list
As part of my overall project, I'm working on some PHP scripts that will do
just that.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Chris Bagnall
-Sent: Friday, October 28, 2005 6:08 AM
-To: 'Asterisk Users Mailing List - Non-Commercial
http://www.asteriskguru.com/tools/bandwidth_calculator.php
Go have a look here and calculate it for yourself.
Zoa
Dan Journo wrote:
Is there an estimate on how many calls a 2Mb ADSL line can handle at
the same time?
Bearing in mind that the upload speed is 256Kb.
Thanks
Dan
Hello All,
Are there any packages need to be installed
before installing the Asterisk.? Cos I am facing problems compiling the
zaptel for the Asterisk... Kindly please do let me know
Regards,
Bharat
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Hello
You cannot have two g729 calls
You can have one g729 and one g711 at same time
regards
Thierry
[EMAIL PROTECTED]
Tel : +33 (0)3 90 40 06 75
Fax: +33 (0)3 90 40 06 76
http://www.widevoip.com
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
We are running 1.0.9 STABLE on all of our machines. I am about try
and upgrade one machine to CVS HEAD as all this echo cancellation
improvements sound enticing. Can anyone recommend
a) A procedure to cleanly upgrade from STABLE to HEAD
b) A procedure to ensure I can back out
Dan Journo a écrit :
Is there an estimate on how many calls a 2Mb ADSL line can handle at
the same time?
It depends on how much hassle you want to put into voice compression.
But without much hassle, using g.729 and SIP or IAX, that's about 10
channels.
Cheers,
Jean-Michel.
Is there an estimate on how many calls a 2Mb ADSL line can
handle at the same time?
Bearing in mind that the upload speed is 256Kb.
Well, on our clients' ADSL connections (256k up and down) we seem to be able
to push between 9 and 12 calls over it with g729 or gsm and iax trunking.
Unless
Bharat M. Sarvan schrieb:
Hello All,
Are there any packages need to be installed before
installing the Asterisk….? Cos I am facing problems compiling the zaptel
for the Asterisk... Kindly please do let me know…
If you're starting with asterisk, you might try [EMAIL PROTECTED] -
Thanks for yourresponse.
Dan
On 28/10/05, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Dan Journo a écrit : Is there an estimate on how many calls a 2Mb ADSL line can handle at the same time?
It depends on how much hassle you want to put into voice compression.But without much hassle, using g.729
Thats very true.
Thanks for pointing that one out.
On 28/10/05, Chris Bagnall [EMAIL PROTECTED] wrote:
Is there an estimate on how many calls a 2Mb ADSL line can handle at the same time?
Bearing in mind that the upload speed is 256Kb.Well, on our clients' ADSL connections (256k up and down) we
Hi
Currently, in
extensions.conf I have:
exten =
_0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})exten
= _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
This enables numbers
to dialledstarting with 0 and 00 and changes them to start with 44.
How can I configure my extensions.conf to
On Thu, 27 Oct 2005, Shane Burrell wrote:
I have a small issue with some remote users connecting to my primary
Asterisk server using 1.0 Every few seconds, there is a subtle tick and a
very small amount of jitter. The tick is not consistent i.e. it could be in
2 seconds, could be 5, could
[EMAIL PROTECTED] wrote :
Boris Bakchiev wrote:
Could echo cancellation on PBX conflict with VPM module and create the
warping babble sound that my users are reporting?
I don't think so, but anything is possible :-)
Do echocancelwhenbridged and echotraining do anything when VPM
I have given up totally on Digium based echo cancel, hardware or
software. The KB1 is the best so far, but still unacceptable. I
installed a hardware echocan FACING the T1 card in the asterisk box, and
all is perfect. No complaints from any of my clients since taking that
leap.
-Darren
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
How can I configure my
extensions.conf to dial a number starting with 44 to dial
without changes? Also a number sent from Outlook starting with +44?
exten =
I would wait for 1.0.0 which is being packaged today, and try it out..
Iqbal
Kanuri, Seshu (Company IT) wrote:
Folks!
I want to know if anyone in the list is using OpenSER,
which appears to be a fork of SER. If so can you post
Your comments on its functionality?
The location where this is
Hi
I have a Sipura SPA 2000 unit and I have configured both the lines in the
unit. both the lines are configured to use 729.
when I make calls from the lines independently it works great. no problem at
all.
when line 1 is connected and when I try to make a call using line 2 while
Hi,
Is anyone running zaptels watchdog in production?
Any adverse effects on using it?
Thanks
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Asterisk-Users@lists.digium.com
Maybe, but I would expect a fax on a Grandstream ATA-286 would be more
reliable than the same fax on the tdm400. I can only speak from my personal
experience. I have faxes setup on both the the Grandstream 286 and on
linksys PAP2NA, with the ATA's on the same 100mbit switch as Asterisk. The
Hello.
I'm having problem with motorola v635 and asterisk. I can make a call
but I can't hear any audio and the other side of the call can hear me
(one way audio).
I'm using usb to bluetooth adaptor (noganet).
I'm using gentoo with kernel 2.6.13-r2, asterisk 1.0.9 and
chan_bluetooth
I was wondering if there is something like that on this Earth:
Some of our users are mobile users - they are rarely in one place for
longer than 15 minutes.
They use mobile phones a lot.
From our mobile operator we have an offer which allows us to call for
free between our mobile phones.
Stay tuned for PhoneCALL's 2.7-RC1 release scheduled soon. We're adding
a new Security Manager that allows you to set the levels of editing for
your users/admins.
Chris Bagnall wrote:
Hello all,
I'm trying to find an Asterisk web interface (or windows gui interface) to
asterisk that won't
Choose Custom install,
and choose Development Tools, Kernel Development Text Based Internet
And also:
OpenSSL-Devel
Readline41
Ncurses4
Ncurses C++ Devel
SOX
Additional Installs
You may also need (depending on your desire to have music on hold) mpg123
which can be found at
Are there any packages need to be installed before installing
the Asterisk.? Cos I am facing
problems compiling the zaptel for the Asterisk... Kindly
please do let me know
Point your web browser to http://www.asterisk.org/download
and read the page carefully.
hello,
I want to learn that, is it 'MUST' to login call queue?
I have 3 call queues, and i want to distribute incoming call to the one
of them. But i don't want to callbacklogin. Because of, after a restart,
all agents have to do callbacklogin.
thanks...
--
Baris Simsek
Project Manager
So the idea is to put a SIM card inside the Asterisk box,
equipped with a special card, a card which would be a mobile
phone really.
There are a number of places that sell GSM gateways (which is what you're
referring to). What I've yet to see are GSM gateways for small business
users that
Hello Everyone!
PhoneCALL version 2.7 http://www.vecsector.com/phonecall is finally
approaching, which will be a major improvement over the past releases
thanks to everyone's input feature requests!
One of the newest features to PhoneCALL is the ability for the entire
interface to be
What about making queuemembers phones instead of agents?
Queues.conf:
[qeuename]
.Blabla.
member = SIP/PhoneName
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Baris Simsek
Sendt: 28. oktober 2005 14:44
Til: Asterisk Users Mailing List -
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
* Pepe Aracil [2005-10-28 10:05]:
I have installed asterisk 1.0.9.dfsg-5 in debian sarge.
if I run /etc/init.d/asterisk stop and then /etc/init.d/asterisk start .
Asterisk don't detach from console where i started it. It beguin to write
all
On Fri, 2005-10-28 at 14:26 +0200, Tomasz Chmielewski wrote:
So the idea is to put a SIM card inside the Asterisk box, equipped with
a special card, a card which would be a mobile phone really.
Does anyone have an idea if such cards exist, and if so, if they work
with Asterisk?
You can
We are looking to acquire E1 service in Fleet right outside of London. I am
in the States so I am not aware of the key players. We currently get ADSL
from Eclipse but were interested in a quote for E1.
What is a typical E1 line go for nowadays and who can I get it from?
Thanks,
Geoff
does anyone know when 1.2 will no longer be beta?
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
yep, thats it.. thank you.
Arne Morten Johansen wrote:
What about making queuemembers phones instead of agents?
Queues.conf:
[qeuename]
.Blabla.
member = SIP/PhoneName
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Baris Simsek
Sendt:
Tomasz,
I'm from Brazil, and we are using here a solution that is based on a box
where we can connect a GSM cellphone and use this directly to a phone or PBX
extension.
I think that you can use some Digium's card (FXS or FXO) on your server,
connect this GSM box there, and route your
You can try DIAX, which I see has some added GSM/PSTN gateway support.
I have yet to try it myself, it looks nice though. -
http://www.laser.com/dante/diax/diaxhlp.htm#gsm
cheersOn 10/28/05, Daniel Varella de Oliveira [EMAIL PROTECTED] wrote:
Tomasz, I'm from Brazil, and we are using here a
Hi all, does anyone know if there is any app/webui that can show
phones that are currently registered to *. I guess this sort of
funcionality counld be grabbed from the CLI with iax2 show peers and
sip show peers, but having little programming knowledge wouldn't know
where to start.
I'm
Get VoiceBlue VoIP GSM gateway.
It works very well with asterisk.
I have been using it for the last 4 month and its fantastic!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Friday, October 28, 2005 10:27 PM
To: Asterisk Users
Hi all, does anyone know if there is any app/webui that can show
phones that are currently registered to *. I guess this sort of
funcionality counld be grabbed from the CLI with iax2 show peers and
sip show peers, but having little programming knowledge wouldn't know
where to start.
I'm
Only the pricing is not that fantastic
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: den 28 oktober 2005 15:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] GSM cards / mobile phone cards
Erick Baum [EMAIL PROTECTED] wrote:
We're having a rather serious echo problem using the Grandstream GXP-2000's
with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking
that might be an easy fix. The echo seems to be worst on internal SIP to SIP
calls but you do get it every once
We use the following device for Asterisk fail-over and our T1s. I
believe they have an E1 version also:
http://www.red-fone.com/fonebridge.html
On Thu, 2005-10-20 at 11:23, John Daragon wrote:
Warning ! I know zip about electronics.
I've been looking for a device to handle the switching of
Erick Baum wrote:
We're having a rather serious echo problem using the Grandstream
GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm
overlooking that might be an easy fix. The echo seems to be worst on
internal SIP to SIP calls but you do get it every once in a while on
You could always (I'll actually do it, I have similar scripts written) just
whip up a php script that connects to a Asterisk Manager Proxy (to limit the
possibility of crashing the server by making too many Manager API
connections), and have it issue the following commands:
Action: Command
Anders Svensson a écrit :
Only the pricing is not that fantastic
It's actually not that bad compared with other GSM gateways.
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Hi all,
through oh323 i can register to my gatekeeper and make
and receive calls.
My gatekeeper routes the incoming call as well as the
outgoing.
The problem is simply that i can't ear nothing from my
SIP ipPhones. I can ear my voice during a call from a
normal telephone in my SIP phone but no
I have an installation with four Qwest POTS lines. For some unknown
reason, Qwest drops the first digit in the dial string, and the call
fails. To fix that problem, I put a 'W' in the dial string:
QWEST=Zap/g2
exten = _9303NXX,1,Dial(${QWEST}/W${EXTEN:1})
The client has since
Daniel Varella de Oliveira schrieb:
Tomasz,
I'm from Brazil, and we are using here a solution that is based on a
box where we can connect a GSM cellphone and use this directly to a
phone or PBX extension.
I think that you can use some Digium's card (FXS or FXO) on your
server, connect
That should be a lowercase w. The placement is correct, just not
a lowercase w.
I have an installation with four Qwest POTS lines. For some unknown
reason, Qwest drops the first digit in the dial string, and the call
fails. To fix that problem, I put a 'W' in the
On Friday 28 October 2005 10:11, Michael Welter wrote:
This works fine for the Qwest line, but Asterisk doesn't absorb the 'W'
for the IAX call--the 'W' is sent as part of the dial string.
Dial(IAX2/${NUMBER:1})
IAX2 isn't limited to numeric numbers. IAX2 can send text, URLs, binary data,
set
persistentmembers = yes
in your queues.conf and the logins of your callback agents
will survive a restart.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Baris Simsek
Sent: Friday, October 28, 2005 2:44 PM
To: Asterisk Users
On Fri, 2005-10-28 at 16:22 +0200, Tomasz Chmielewski wrote:
looks interesting.
do you know by chance how much such a single-cell box cost (more or less)?
I found it here http://www.thehightechstore.com/plugcell.html
at 295$USD
--
Dave Cotton [EMAIL PROTECTED]
Geoff Manning wrote:
We are looking to acquire E1 service in Fleet right outside of London. I am
in the States so I am not aware of the key players. We currently get ADSL
from Eclipse but were interested in a quote for E1.
What is a typical E1 line go for nowadays and who can I get it from?
On Fri, Oct 28, 2005 at 04:40:07PM +0200, Dave Cotton wrote:
On Fri, 2005-10-28 at 16:22 +0200, Tomasz Chmielewski wrote:
looks interesting.
do you know by chance how much such a single-cell box cost (more or less)?
I found it here http://www.thehightechstore.com/plugcell.html
at 295$USD
I'm getting the following error when starting Asterisk: Error while writing
audio data: broken pipe. In my processesses I have tons of mpg123 instances
running, probaby because of asterisk trying to start ad nauseum.
What could be creating this? I am running Beta 1.2, trying to see if
I'm getting the following error when starting Asterisk: Error while writing
audio data: broken pipe. In my processesses I have tons of mpg123 instances
running, probaby because of asterisk trying to start ad nauseum.
What could be creating this? I am running Beta 1.2, trying to see if
Thanks for that Adam, fantastic!
I did need to add one line to get it to work
#!/usr/bin/perl
#
##get lists of registered peers from asterisk
$iaxpeers = `/usr/sbin/asterisk -rx \iax2 show peers\`;
$sippeers = `/usr/sbin/asterisk -rx \sip show peers\`;
##replace newline characters with html
Hi folks,
Is anyone aware of a way to prevent transcoding or better still apply
some kind of weighting to codec selection based on other channels in the
call? Let's say we support g729 and gsm, a peer supports both and a
client supports one of them. We're seeing calls frequently coming in on
Hello!
I have installed 2 servers, one with SER integrated
with PostgreSQL (Fedora Core 3)andthe other withAsterisk
(Fedora Core 4). I can talk Softphone - SER - SER - Softphone (in
caseI try to contact aperson that as a different SIP server). Now
the goal is to, use Asterisk as a gateway,
I'm trying to configure an Asterisk server to make inbound and
outbound calls through a Zultys MX250 operating as a SIP media gateway.
This is my first experience with Asterisk, but from what I
understand, I need to register the Asterisk system as a SIP device
with the MX250. This is what
Hello All ,
On Thu, 27 Oct 2005, Mr. James W. Laferriere wrote:
On Thu, 27 Oct 2005, Phil Pritchard wrote:
only new to asterisk, but have had some hardware exp.
stay away from irq9 its tied to irq2 and will always be shared, Paul has
the go.. in bios disable serial and or
Can anyone out there help me ?
Beside countless benefits, without Caller-ID on my
SIP Phone's screen * is a real pain in my neck.
I have * 1.2.0-Beta (Latest CVS) and a X100P card.
Sometimes I can get the Caller-ID and sometimes I can
not. Even from the same number. Any suggestions ?
Mojo with Horan Company, LLC wrote:
The recent suggestion on the list was to not use 1.0.9 zaptel
You mean the driver, or the version of fxotune? fxotune has been removed
from the prior versions of the zaptel driver, it's only included in 1.2
now. As for the driver, is anyone using the 1.2
I've got a cell phone setup as an extension in a queue. On occasion the
cell phone will drop the call due to loss of, or bad, signal. Is there a
clean way in the dial plan to reintroduce a call back into the queue
when the call is dropped on the extension side? I realize this would
occur
We had to move from a old * server to a new one in a hurry (hardware
failure). The old server was a dual pentium 700 with 512MB ram running
fedora core 2, the new one is a single 3GHz Pentium with 1gb ram.
The same number of people are connected to the new server as the old,
the same number
Thank you Hadley, that was the problem.
Cheers,
Richard
- Original Message -
From: Hadley Rich [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, October 28, 2005 12:20 AM
Subject: Re: [Asterisk-Users] Not saving
On Wed, Oct 26, 2005 at 10:12:09AM -0400, Matt wrote:
Does anyone know if SIPURA SPA-2002's support DNS SRV records?
Yep, it does (as does its brother PAP2-NA).
Ray
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Darren,
Can you elaborate on what echocan did you use and how?
Thanks.
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Wright
Sent: Friday, October 28, 2005 7:35 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
snip
Well, the major incumbent is BT.
Are you sitting down ?
Installation :
Per channel 1 year contract 3/5y contract 3/5y+commitment
First 15 channels (min 8)GBP 125 GBP 80GBP 0
16-30 (per channel) GBP 30 GBP 15GBP 0
If anyone knows of smaller-scale units that work on GSM900 and 1800, I'd
also love to hear about them.
You might want to investigate a Nokia 22
(http://europe.nokia.com/nokia/0,8764,56024,00.html). This provides a single
GSM line which is interfaced to the PBX by an anlogue trunk/extension.
On Fri, Oct 28, 2005 at 05:40:05PM +0100, Charles Trevor wrote:
Well, the major incumbent is BT.
Are you sitting down ?
Installation :
Per channel 1 year contract 3/5y contract 3/5y+commitment
First 15 channels (min 8)GBP 125 GBP 80GBP 0
16-30 (per
On Fri, Oct 28, 2005 at 09:59:00AM +0200, Pepe Aracil wrote:
Hello.
I have installed asterisk 1.0.9.dfsg-5 in debian sarge.
Did you install the binary package from unstable or rebuilt it?
if I run /etc/init.d/asterisk stop and then /etc/init.d/asterisk start .
Asterisk don't detach from
On Fri, Oct 28, 2005 at 05:43:03PM +0100, David Cook wrote:
You might want to investigate a Nokia 22
(http://europe.nokia.com/nokia/0,8764,56024,00.html). This provides a single
GSM line which is interfaced to the PBX by an anlogue trunk/extension. From
memory they cost around £100-150. I
On Fri, Oct 28, 2005 at 09:39:59AM -0400, Adam Moffett wrote:
Hi all, does anyone know if there is any app/webui that can show
phones that are currently registered to *. I guess this sort of
funcionality counld be grabbed from the CLI with iax2 show peers and
sip show peers, but having
Hi everyone!
I just got a zultys zip 2 today with no manuals. Can anyone tell me how
to get in and config the=is thing please? I know there has to be some
*super secret code* to enable dhcp on it somehow and then a login as
password for the web interface or something? HELP!!???
--
-Linc
It costs here more or less R$600,00 (about US$264,55)
Our friend, Dave Cotton post a message with a good price for outside of
Brazil. US$295,00 is a good price, I think.
I know that guy in Sao Paolo (the correct is São Paulo), that the site
http://www.thehightechstore.com/plugcell.html
Hello Folks,
I thought Id make a sorta announcement as Ill be
in Montreal on
a partial vacation/partial hangout/partial meet and greet thing. I thought it
might be nice for all the people in the area, and perhaps those attending the
Meet Asterisk thing to get together for supper and
I use the same type of thing in my PHP without going through the Manager
port:
I had to do chmod u+s /usr/sbin/asterisk in order for my apache server
to be able to connect and get the response...
Thanks for that Adam, fantastic!
I did need to add one line to get it to work
#!/usr/bin/perl
We have 50 of these phones in one location and a couple remote phones. The problem seems to be caused by the volume settings on the phone. We have noticed that the echo seems to be worse when the volume is very high on the phone (not using speakerphone). We're still testing, but that's what we've
Does sound like you have the fix - upgrade to a newer Asterisk.
*groan* Yes, it did solve the problem, 100%. I upgraded a single site to
1.0.9 and call quality is perfect. Now, on to the other 29thank GOD for
SSH.
___
--Bandwidth and Colocation
Don't thank me, it's Mgernoth and kb1_kanobe that get the props for all
of this. They've been doing a lot of work to improve the software echo
cancelers lately.
Matthew Fredrickson
On Oct 28, 2005, at 1:29 AM, [EMAIL PROTECTED] wrote:
Hello Matthew,
It is always nice to see improvements.
On 10/28/05, Dustin Wildes [EMAIL PROTECTED] wrote:
Stay tuned for PhoneCALL's 2.7-RC1 release scheduled soon. We're adding
a new Security Manager that allows you to set the levels of editing for
your users/admins.
Chris Bagnall wrote:
Hello all,
I'm trying to find an Asterisk web
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