Matt wrote:
--
What this means is that if the call is busy, it will play busy tones,
if the call is ringing it will play ringing, congestion, congestion
etc.
The reason you are hearing silence is that Asterisk doesn't know what
the status of the call is before
Matt Riddell wrote:
On 6/11/2008 8:37 p.m., Thomas Kenyon wrote:
Jeff LaCoursiere wrote:
I didn't realize only 4% of the world's population lived in North America!
Learn something every day.
Sorry that was my bedtime maths, the figure is just over 4.5%.
4.5611893661578069635904176186202%
Dear list
anyone know wich is the limit of maxload into asterisk.conf ?
Also the meaning is related to RAM ? or CPU ?
Regards Andrea
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On Thu, Nov 06, 2008 at 08:42:51AM +, Thomas Kenyon wrote:
Matt Riddell wrote:
On 6/11/2008 8:37 p.m., Thomas Kenyon wrote:
Jeff LaCoursiere wrote:
I didn't realize only 4% of the world's population lived in North
America!
Learn something every day.
Sorry that was my bedtime maths,
Thomas Kenyon wrote:
Or to be outright pedantic 4.5380853065046927068273204778542%.
I apologise for attaching the files, It was unintentional.
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BJ Weschke wrote:
Ade Vickers wrote:
-Original Message-
Hi Folks,
I'm using the rawplayer program to provide my
music-on-hold, and it
works very well, with one small
drawback: every time I reset Asterisk, for any reason, the
MoH resets
to the beginning of the list.
Hi,
This week's VoIP Users Conference will be tie the Talkshoe PSTN/SIP
ULAW conference bridge to the ZipDX.com G.722-capable bridge. It may
be a little crazy, but I'm looking forward to getting some
explanations from David Frankel about the effects of wide band (or as
Polycom calls it, HD Voice)
didforsale.com have just sent me SPAM to the email address I use here.
What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that
I'll never used their services. Morons.
Gordon
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On Thu, Nov 06, 2008 at 07:26:06AM +, Gordon Henderson wrote:
Are you using a ring-adapter for the UK that includes the capacitor to put
the ringing current on pin 3? Something like this:
http://www.voipon.co.uk/rj11-adaptor-with-ring-capacitor-p-278.html
Or take the output of the
Gordon Henderson wrote:
didforsale.com have just sent me SPAM to the email address I use here.
What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee that
I'll never used their services. Morons.
Likewise.
Bails
Gordon
___
Hi Steve.
I'm still trying the same because I'm interested in the subject.
For what I can understand the ExtenSpy application is working properly
if the selected extension receives a call. Seems not working, instead,
if the selected extension originates the call.
My actual setup is like that:
Hello,
Today I saw about 40 calls drop on my asterisk box. Its doing Zap to SIP w/
g729 compression. Wasnt sure what the problem is and now I'm monitoring the
console and I see these strange errors. I'm running Asterisk 1.2.24
Nov 6 05:09:09 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk configuration files support #include and a number of other
interesting tricks, as mentioned in
Hi,
We have an issue where Polycom's lose BLF functionality after a reboot. The
only way to fix it is to reboot the Polycoms.
Anyone else have this issue? We are using 1.4.18.
If I run 'sip show subscriptions' all the subscriptions come back after the
restart but the lights on the phones
Mr Shunz wrote:
Hi,
We have an issue where Polycom's lose BLF functionality after a reboot. The
only way to fix it is to reboot the Polycoms.
Anyone else have this issue? We are using 1.4.18.
If I run 'sip show subscriptions' all the subscriptions come back after the
restart but the
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
read(0, unfinished ...
+++ killed by SIGSEGV (core dumped) +++
Process 15755 detached
On a second
Use snom M3 Siemens got some problems.
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Louis-David Mitterrand wrote:
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
Fresh install or upgrade?
If it was over the top upgrade, it
What about Mexico and Canada? Aren't they considered North America?
j
On Thu, 6 Nov 2008, Thomas Kenyon wrote:
Matt Riddell wrote:
On 6/11/2008 8:37 p.m., Thomas Kenyon wrote:
Jeff LaCoursiere wrote:
I didn't realize only 4% of the world's population lived in North America!
Learn
My guess is that anything part of NAFTA is considered North America.
It is kind of strange how a county with such a small percentage of the world
population holds pretty much the entire world's markets and economy in it's
hands.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212
On Wed, 5 Nov 2008, Pedram M wrote:
Any recommendations on good wireless SIP phones?
VoIP Tech Chat did a review on the Linksys WIP 330:
http://tinyurl.com/review330
and VoIP Supply has a new phone (haven't read any reviews) that has a
new long-life battery.
Fred Posner
smime.p7s
On Thu, Nov 6, 2008 at 4:56 AM, Pedram M [EMAIL PROTECTED] wrote:
Any recommendations on good wireless SIP phones?
I use a Siemens S675IP in our two person office. It performs very
well, and has a built in answering machine which is of interest for us
because we have several SIP accounts that
Hi,
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4 min)
This is triggered by the monitoring app sending a POKE to the iax port.
The warning appears
Hi,
I'm trying to make odbcexec work with Asterisk 1.6.
I had the attached code (app_odbcexec, not the standard one) working great
with asterisk 1.2 an MSSQL Server on heavy load PBXs with no problem, I'm
trying to port this to asterisk 1.6 but I'm failing to do so.
I attach de working code in
On Thu, 6 Nov 2008 15:01:09 +0100, randulo wrote:
On Thu, Nov 6, 2008 at 4:56 AM, Pedram M [EMAIL PROTECTED] wrote:
Any recommendations on good wireless SIP phones?
I use a Siemens S675IP in our two person office. It performs very
well, and has a built in answering machine which is of interest
2008/11/5 Atis Lezdins [EMAIL PROTECTED]
On Wed, Nov 5, 2008 at 5:28 PM, Olivier [EMAIL PROTECTED] wrote:
2008/11/5 Atis Lezdins [EMAIL PROTECTED]
On Wed, Nov 5, 2008 at 12:39 PM, Olivier [EMAIL PROTECTED] wrote:
Hi,
I've new to http://www.voip-info.org/wiki/view/Asterisk+AEL2
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk configuration files support #include and a number of other
On Thu, Nov 6, 2008 at 3:21 PM, Michael Graves [EMAIL PROTECTED] wrote:
The S675/685IP supports G.722 which is great! But it has no mute
button, which is a drag. Also, its less expensive.
Truth be told, I hate that there's no mute button. Also, the handset
isn't good enough to make a huge
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4 min)
This is triggered by the monitoring app sending a POKE to the
On Thu, 2008-11-06 at 12:16 -0200, Sebastian Gutierrez wrote:
I'm trying to make odbcexec work with Asterisk 1.6.
Why not just use the functionality of func_odbc already built into
Asterisk 1.6? Is there something you gain by going with odbcexec that
func_odbc doesn't provide?
Also, just as a
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15] WARNING[2209]: chan_iax2.c:7000 socket_process:
midget packet received (1 of 4
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk configuration files support #include and a number of other
On Thursday 06 November 2008 08:53:40 Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Louis-David Mitterrand wrote:
When monitoring an asterisk through its iax2 port I get these warnings
at the console:
[Nov 6 13:15:15]
Ok, sorry for the response on the same thread.
The main thing is that with this I set the Store Procedure or Query directly
on the dialplan line, is easier to configure, change, manage, etc.
I also know that works great with heavy load, and it reconnects when the
network goes down and up.
Can you
Local channel will help you send your call through the dialplan.
You can make all your decision there.
If it answers, then the specified application will be execute.
Check this example
http://www.astblog.com/2008/09/18/use-the-power-of-local-channels/
David Klaverstyn wrote:
I have
Hello !
I am experiencing some problems with Asterisk trunking, this is the scenario:
There are 3 servers, a DID server provider (VOIP provider) which
delegates us a bunch of DID numbers to our asterisk server number one
(I will call it AA), from which I route the calls to Asterisk server
number
On Thu, 6 Nov 2008, Gordon Henderson wrote:
didforsale.com have just sent me SPAM to the email address I use here.
What a bunch of stupid fuckwits. How to get a 100% cast-iron guarantee
that I'll never used their services. Morons.
The English have such a way with words :)
I keep a local
It sounds like you have analog lines. If that is the case, the silence
you experience is Asterisk sending the DTMF down the line. Asterisk
collects the DTMF and when you are done dialing it retransmits those
digits down the analog line. I think each digit is by default 300ms.
If you are
I'm glad I'm not the only one who got that. I sent them a nasty response
earlier this morning...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards
Sent: Thursday, November 06, 2008 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two offices are separate companies but support
one another and
I sent this email a few days ago but did not see any responses to it:
I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for
testing. In addition I register a zoiper SIP soft phone.
For the Grandstream I have busylevel=1 in sip.conf.
If I place a call from the GXP280
On Thu, Nov 6, 2008 at 2:11 PM, David Gibbons [EMAIL PROTECTED]wrote:
I'm glad I'm not the only one who got that. I sent them a nasty response
earlier this morning...
I got the same crap from them. I can't imagine anyone buying from a company
that spams subscribers of a mailing list to get
On Thu, Nov 6, 2008 at 6:12 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a
Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
to completely fail, then there won't be any spam.
David Gibbons wrote:
I'm glad I'm not the only one who got that. I sent them a nasty response
On Thursday 06 November 2008 10:12:11 Brent Davidson wrote:
If I have a global variable in my dialplan and I change it, does that
change immediately take affect for all calls that are active?
Here is my situation. The company I work for has two office groups that
share a building. The two
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
to completely fail, then there won't be any spam.
Socialism has already completely failed.
Tilghman Lesher wrote:
[companyA]
exten = _X.,1,Set(company=A)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
[companyB]
exten = _X.,1,Set(company=B)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
I should probably also mention that I am using AEL for my dialplan.
(i'm a programmer and the
Wi-Fi SIP phones aren't limited to hot spots. I am in the process of setting
up asterisk for SOHO. At present I'm not even using VoIP trunking, only LAN to
stns and I intend to use Wi-Fi instead of analog cordless phone. I got the
Engenius one, and it works, but I haven't played with it
http://en.wikipedia.org/wiki/Jacque_Fresco
A resource based economy.
Greg Woods wrote:
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
If it is 300 ms, that is way to long. I don't know any CO grade receiver that
can't decode in 80 ms and some can do 40. There is also a similar size gap
between digits.
Is there an option to start dialing as soon as enough digits are collected to
guarantee a unique route? That has been the
On Thursday 06 November 2008 11:41:08 Brent Davidson wrote:
Tilghman Lesher wrote:
[companyA]
exten = _X.,1,Set(company=A)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
[companyB]
exten = _X.,1,Set(company=B)
exten = _X.,n,Goto(maincontext,${EXTEN},1)
I should probably also mention
On Thu, Nov 6, 2008 at 7:50 PM, Anthony Francis [EMAIL PROTECTED] wrote:
http://en.wikipedia.org/wiki/Jacque_Fresco
A resource based economy.
Greg Woods wrote:
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
Gotta love this list being farmed for spammers now. I am sure they call
Eric wrote:
It sounds like you have analog lines. If that is the case, the silence
you experience is Asterisk sending the DTMF down the line. Asterisk
collects the DTMF and when you are done dialing it retransmits those
digits down the analog line. I think each digit is by default 300ms.
--Original Message Text---
From: Wilton Helm
Date: Thu, 6 Nov 2008 10:34:35 -0700
Wi-Fi SIP phones aren't limited to hot spots. I am in the process of
setting up asterisk for SOHO. At present I'm not even using VoIP
trunking, only LAN to stns and I intend to use Wi-Fi instead of analog
cordless
Jeff LaCoursiere schrieb:
What about Mexico and Canada? Aren't they considered North America?
Canada: yes. Mexico: depends on the definition I guess.
http://en.wikipedia.org/wiki/North_America#Countries_and_territories
Philipp Kempgen
--
http://www.das-asterisk-buch.de -
I think I'll take the occasional spam and keep my freedoms and civil
liberties...
Tell Kim Jong Il I said hello though!
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Thursday, November 06, 2008 11:46 AM
To: Asterisk Users Mailing
He's dead, if you look at the recent photos of him his shadow is not
where it should be compared to other people in the photos. Its all
photoshop'ed now :)
David Gibbons wrote:
I think I'll take the occasional spam and keep my freedoms and civil
liberties...
Tell Kim Jong Il I said hello
Ok, sorry for the response on the same thread.
This is a new one.
The main thing is that with this I set the Store Procedure or Query directly
on the dialplan line, is easier to configure, change, manage, etc.
I also know that works great with heavy load, and it reconnects when the
network
Steve Totaro schrieb:
It is kind of strange how a county with such a small percentage of the world
population holds pretty much the entire world's markets and economy in it's
hands.
pretty much the entire is not entirely true. :-)
On Thursday 06 November 2008 12:27:05 Sebastian Gutierrez wrote:
The main thing is that with this I set the Store Procedure or Query
directly on the dialplan line, is easier to configure, change, manage, etc.
I also know that works great with heavy load, and it reconnects when the
network
Greg Woods wrote:
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
to completely fail, then there won't be any spam.
Socialism
On Thu, 2008-11-06 at 13:55 +0100, Olivier wrote:
Yes, you're right : NoOp needs verbosity of 3 and above.
Thanks for helping.
The surprising thing is that AEL Verbose prints output whatever the
verbosity level is (even with 0).
Would you qualify this as normal ?
Olivier--
The
Singer X.J. Wang wrote:
He's dead, if you look at the recent photos of him his shadow is not
where it should be compared to other people in the photos.
Well that's just lovely. Kim Jong Il is now an immortal vampire.
Better call the white house and tell them to replace the nuclear warhead
On Thu, 2008-11-06 at 17:02 +0200, Tzafrir Cohen wrote:
On Thu, Nov 06, 2008 at 12:57:15PM +0200, Tzafrir Cohen wrote:
Hi
I'm lately bothered with the need to provide a set of Asterisk
configuration files in a package that will be good for a wide range of
Asterisk users.
Asterisk
Most IVRs want longer DTMF tone lengths. If you shorten the
toneduration= then many IVRs won't work.
Wilton Helm wrote:
If it is 300 ms, that is way to long. I don't know any CO grade receiver
that can't decode in 80 ms and some can do 40. There is also a similar size
gap between digits.
LOL, I love people like Jacque who have no clue that wolves exist. The assumption that evil does not exist in the hearts of men is niavity.
What this discussion has to do with asteriskI have no clue... but entertaining none-the-less.
http://en.wikipedia.org/wiki/Jacque_Fresco
A resource
I had problems when I was playing with the ExtenSpy command as well. The
issue for me was that the context for the extension that I was using was not
the same as the one that Asterisk showed in the console output when I called
the phone. This is because I have various contexts included in other
The linksys phones annoy me because they cannot implement southern
hemisphere DST properly. Grr.
(yes, you can do it with a hack - but why can't the phones just work?)
PaulH
Steve Anness wrote:
Good Day,
I have been tasked with fixing the time on our asterisk server. I am
having a hard
The linksys phones annoy me because they cannot implement southern
hemisphere DST properly.
I was shocked the first time I had to write firmware for an international
project. Not only is there the southern hemisphere issue of opposite seasons,
but just about anyone in the world with a
I found out what the problem was.
It appears to be a bug in the Polycom 430 firmware.
I have 2 lines on the phone and both of them use the same auth id but with
different servers.
It seems that if you make an outgoing call from the phone on line 2 and then
called party hangs up. Asterisk says
Dou you have any example? Can I call directly to querys without the
templates???
Thanks!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Tilghman
Lesher
Enviado el: Thursday, November 06, 2008 4:53 PM
Para: Asterisk Users Mailing List - Non-Commercial
What happens?... I'm not sure. IAX peers work fine, but SIP users does not
register.
There are not firewalls blocking ports.
But actually the problem is not the issue because I tried with normal kernel
and doesn't work. It is not configuration because it worked on a virtual
machine on VirtualBox.
On Thursday 06 November 2008 18:59:29 Sebastian Gutierrez wrote:
Dou you have any example? Can I call directly to querys without the
templates???
func_odbc.conf:
[EXEC]
read=${ARG1}
write=${ARG1}
dsn=something
extensions.conf:
exten = 123,1,Set(result=${ODBC_EXEC(SELECT foo FROM bar)})
--
Hi All,
I need a help on g729 codec.Is there any tool which can convert g711 codec into
g729 codec and supports batch processing ?
Thanks in advance
vivek
--- On Fri, 11/7/08, Edgar Guadamuz [EMAIL PROTECTED] wrote:
From: Edgar Guadamuz [EMAIL PROTECTED]
Subject: Re: [asterisk-users]
On Thursday 06 November 2008 23:45:38 vivek rastogi wrote:
I need a help on g729 codec.Is there any tool which can convert g711 codec
into g729 codec and supports batch processing ?
for from in /full/path/to/directory/*.ul ; do
to=${from%%.ul}.g729
asterisk -rx file convert
Hello,
Has anyone got any ideea if I can use in Asterisk the new called party
number optionally included in the diagnostic field for release cause 22
in ISDN?
A callcenter gets lots of this messages from the telco and it would be
nice if I could tell them that the number has
All of you on this list are familiar with how DNS works. You probably
use spam blocking lists (SBL) for your email servers? In case someone
is interested in building accurate SBL, Spamcop is a service that
allows you to easily report spam by sending in the headers. It can be
automated by procmail
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