On Thu, 26 Feb 2009, Haim Dimer wrote:
Hello,
Some of my users have phones lacking a DND button. I need to provide
an extension they can dial that will put them in DND, i.e. tell the
server not to send them any calls until they get off the DND.
I've researched it for almost 3 days now and
Hey, thanks for the help David, Tzafrir.
Lots of config tips there :-)
We managed to find a fix through the following (For anyone who's
interested):
Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an
error of:
Notice: Configuration file is /etc/zaptel.conf
line
2009/2/27 Bill Michaelson b...@cosi.com
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
Afaik only by limiting the number of call files in the directory.
___
-- Bandwidth and
Hi,
I am trying to log queue_log to odbc (MS SQL) I have res_odbc.conf
configured and modules.conf have
preload = res_odbc.so
preload = res_config_odbc.so
extconfig.conf has queue_log = odbc,asterisk.
When I start asterisk I get the following messages. The important one being:
Realtime
paste your sip.conf.
David
2009/2/26 michel freiha mich...@gmail.com
Dear All,
I have created an inbound context in SIP .conf that forward incoming call
to opensips server...The problem appears as soon as I enable t38pt_udptl =
yes...The Asterisk negotiate the SIP session with OpenSIPS
Dear David,
Please find on http://pastebin.com/m69b8559d my sip.conf file
Thanks a lot
On Fri, Feb 27, 2009 at 1:05 PM, David fire ddf...@gmail.com wrote:
paste your sip.conf.
David
2009/2/26 michel freiha mich...@gmail.com
Dear All,
I have created an inbound context in SIP .conf that
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'
We then Chmodded everything under /dev/zap/ , rebooted and almost fell off
our chairs when it worked!
By right, if the problem is due to this error, you should see a permission
error message in
FYI to everyone...
It was an issue on Vitelity's end on the gateway I was assigned to. They
switched me, and it's working fine now.
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
I turned on DTMF debugging. It looks like the extra digits coming in
are less than
On Fri, Feb 27, 2009 at 07:12:41PM +0900, Wye-khe Kwok wrote:
We managed to find a fix through the following (For anyone who's
interested):
Running /sbin/ztcfg -vv to configure Zaptel initially resulted in an
error of:
Notice: Configuration file is /etc/zaptel.conf
line 0:
Hello
OK I have tried this in my dialplan:
exten = _0262XX,1,Set(GROUP()=Reunion)
exten = _0262XX,2,GotoIf(${GROUP_COUNT(Reunion)} 24 ? 500)
exten = _0262XX,n,NoOp(This channel is member of group: ${GROUP()})
exten = _0262XX,n,NoOp(Number of channels is ${GROUP_COUNT(Reunion)})
Hi,
I have problem with Asterisk 1.6.0.1. I need to change language for
playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime
plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a
bug ...? So I paste my test dialpan and prompt's locations. I hope this
The correct syntax for GotoIf is:
exten = _0262XX,n,GotoIf($[${GROUP_COUNT(Reunion)}24]?500)
Otherwise it seems to evaluate the string number 24 which is always
true.
Duh...
Thx
JM
--
Jean-Michel Hiver - Synapse co-founder CTO
GSM +262 692 828 070
Hello List,
The list sorted my problem thus I shall contribute back ;-)
PROBLEM:
I am posting this example, where I have a Reunion link of 30 channels. If
i send all the traffic (proper + mobile) on the link, the less profitable
proper traffic fills the link and leaves no channel for
On Feb 26, 2009, at 3:05 PM, Tzafrir Cohen wrote:
On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote:
At the top of my /etc/dahdi/system.conf file is this line:
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Feb 25
18:25:10 2009
-- do not hand edit
OK, so how do I adjust
Hi folks
A common wisdom here is that one should use a proper hardware phone
rather that an extra software on the user's PC. Why is that such a big
issue?
One thing that bothers me with the current crop of hardware SIP phones
is that they are hopelessly properitary.
So what would it take to
On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote:
Gavin Henry wrote:
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of Asterisk-gui/FreePBX/a-n-other gui?
I've read lots
Hi Paulo,
It's right! I've changed the zend.zel_compatibility_mode to Off,
following your suggestion, and asterisk-stat is still working on PHP5.
Thank you!
Just for clarity: the default values for the two keys on OpenSuse 10.2
(updated to latest revision), and following, is Off.
Best regards,
Tzafrir Cohen wrote:
Hi folks
A common wisdom here is that one should use a proper hardware phone
rather that an extra software on the user's PC. Why is that such a big
issue?
One thing that bothers me with the current crop of hardware SIP phones
is that they are hopelessly properitary.
Tzafrir Cohen wrote:
Hi folks
A common wisdom here is that one should use a proper hardware phone
rather that an extra software on the user's PC. Why is that such a big
issue?
Marketability for one. People worldwide understand the telephone
paradigm. You have a handset and a box with
1) How can I use codec_dahdi? Would it be useful when passing a call from
one dahdi channel to another dahdi channel?
It is used whenever you need G729 or G723 transcoding (or any other
format supported by the Digium transcoding board). If you don't have a
Digium transcoding board then you don't
Just a tip: throw extensions.conf away and use extensions.ael - much
more easy:
_0262XX = {
Set(GROUP()=Reunion);
if( ${GROUP_COUNT(Reunion)} 24) {
NoOp(Total channels congested, retuning NOCAV);
Congestion();
} else {
NoOp(This channel is member of group: ${GROUP()});
It's a dream!
It's since years that I'm thinking to have an open hardware project
targeted to a SIP application.
I'm thinking, for example, to have a modular system that can be targeted
to different custom appliances like, for example, (video) door bell
opener/intercom, or building/desktop music
Marco Signorini wrote:
It's a dream!
It's since years that I'm thinking to have an open hardware project
targeted to a SIP application.
there is already a project called openmoko - join it and buy some hardware.
The phone is large and clunky - the idea is good, but not something
you're
Jon Pounder wrote:
Marco Signorini wrote:
It's a dream!
It's since years that I'm thinking to have an open hardware project
targeted to a SIP application.
there is already a project called openmoko - join it and buy some hardware.
The phone is large and clunky - the idea is
I don't think they are locking the same device that you buy when you buy
the -NA version. I believe that Linksys is making pre-configured
devices for these large buyers and selling them much cheaper to them in
bulk than they sell the -NA version to the community at large.
I'm sure you are
Thanks for your reply,
On Fri, Feb 27, 2009 at 10:53:21AM -0500, SIP wrote:
Tzafrir Cohen wrote:
Hi folks
A common wisdom here is that one should use a proper hardware phone
rather that an extra software on the user's PC. Why is that such a big
issue?
Marketability for one.
I assume that the relevant application requires some non-trivial CPU power. I
would
exclude e.g. a 486-based systems.
I'm not sure that's the case. The industry has gone in the direction of
throwing lots of silicon at a problem, often as an excuse for poorly written
code, sometimes in an
Wilton Helm wrote:
I assume that the relevant application requires some non-trivial CPU
power. I would
exclude e.g. a 486-based systems.
I'm not sure that's the case. The industry has gone in the direction
of throwing lots of silicon at a problem, often as an excuse for
poorly
On Fri, 27 Feb 2009, Tzafrir Cohen wrote:
Hi folks
A common wisdom here is that one should use a proper hardware phone
rather that an extra software on the user's PC. Why is that such a big
issue?
Both my laptop and desktop PC have poor quality microphone inputs. I use a
USB phone on my
2009/2/27 Wilton Helm wh...@compuserve.com
I assume that the relevant application requires some non-trivial CPU
power. I would
exclude e.g. a 486-based systems.
I'm not sure that's the case. The industry has gone in the direction of
throwing lots of silicon at a problem, often as an
On Fri, Feb 27, 2009 at 09:40:09AM -0700, Wilton Helm wrote:
I assume that the relevant application requires some non-trivial CPU power.
I would
exclude e.g. a 486-based systems.
I'm not sure that's the case. The industry has gone in the direction
of throwing lots of silicon at a
Two things
The dev/zap problem was probably fixed by a modprobe that occurred on
the reload and therefore had no relevance to the chmod.
Ztdummy is created by zaptel and used in some non-analog functions
AFAIK
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Some variant of the ulimit command would accomplish this but YMMV and
Caveat Emptor.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill
Michaelson
Sent: Thursday, February 26, 2009 7:59 PM
To:
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
On Fri, 27 Feb 2009, Danny Nicholas top posted:
Some variant of the ulimit command would accomplish this but YMMV and
Caveat Emptor.
Which one?
-fs::sedwards:~$ ulimit
I am doing:
Action: Monitor
File: /tmp/testing
Format: gsm
then Mix:0 only records the output
Mix: 1 combines in the input and output
I wish (in this case) to only record the input - how do I do this?
Jerry
___
-- Bandwidth and Colocation Provided by
On Fri, 27 Feb 2009, Tzafrir Cohen wrote:
The core of the system is:
http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp
(yes, also two TDM ports with slics. No idea how to use them)
That plug supports running quite a number of Linux distributions
(Debian, Fedora, Gentoo,
You specify a channel. Just specify the channel on the other leg of the
call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Jerry Geis ge...@pagestation.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date:
On Fri, Feb 27, 2009 at 05:48:28PM +, Gordon Henderson wrote:
On Fri, 27 Feb 2009, Tzafrir Cohen wrote:
The core of the system is:
http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp
(yes, also two TDM ports with slics. No idea how to use them)
That plug
Hi,
I have a growing voip business Im i looking a solution that can handle at
least 3000-4000 concurrent calls
with great performance. Also with a billing platform, reports, reseller
platform, LCR, call routing,real time reports, SQL dababase access
real time Load Reports.
Any recommendation?
Gordon Henderson wrote:
On Fri, 27 Feb 2009, Tzafrir Cohen wrote:
The core of the system is:
http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp
(yes, also two TDM ports with slics. No idea how to use them)
That plug supports running quite a number of Linux
Again, the main reason for me to require a higher end CPU is audio
compression. But I also want the system to be run by a standard OS. It
needs to be easy to add your own application there.
Mutually exclusive. I don't know any standard OS that doesn't waste about 10 x
as many CPU cycles as a
Tzafrir Cohen wrote:
On Fri, Feb 27, 2009 at 05:48:28PM +, Gordon Henderson wrote:
On Fri, 27 Feb 2009, Tzafrir Cohen wrote:
The core of the system is:
http://www.marvell.com/products/embedded_processors/kirkwood/index.jsp
(yes, also two TDM ports with slics. No idea how to use
Wilton Helm wrote:
Again, the main reason for me to require a higher end CPU is audio
compression. But I also want the system to be run by a standard OS. It
needs to be easy to add your own application there.
Mutually exclusive. I don't know any standard OS that doesn't waste
about 10 x
On Fri, Feb 27, 2009 at 01:07:57PM -0500, Jon Pounder wrote:
this mentality mystifies me too - why would you want a pbx running on a
handset ? ok so you can, but once that novelty wears off in 5secs what
is the practical use ? The only reason would be if you don't have any
other phones,
On Fri, Feb 27, 2009 at 11:11:35AM -0700, Wilton Helm wrote:
Again, the main reason for me to require a higher end CPU is audio
compression. But I also want the system to be run by a standard OS. It
needs to be easy to add your own application there.
Mutually exclusive. I don't know any
Here is a link to a better, but possibly dangerous answer.
http://www.netadmintools.com/art295.html
Since a typical linux box probably allows about 250K files to be
simultaneously open, and you need about 2K for system and * overhead, by
cutting the max number of files down to about 3K, you
This is not entirely true - many of the nokia phones use a java OS as a
core, and you can load pretty much any java software you want on them,
but all the points about power and battery use are still valid. (and
whether you really consider that truly an OS is questionable, but its
out there)
Set the ctime of the spool file in the future and Asterisk will not
process the file until that time.
Danny Nicholas wrote:
Here is a link to a better, but possibly dangerous answer.
http://www.netadmintools.com/art295.html
Since a typical linux box probably allows about 250K files to be
Is there a way to force a channel to continue in the dialplan after
the remote end hangs up?
Specifically, I am trying to play around with setting up a fax
server. I can receive the fax, but sometimes the sending fax hangs up
before my System command for printing can run and the fax never
Daniel Hazelbaker wrote:
Specifically, I am trying to play around with setting up a fax
server. I can receive the fax, but sometimes the sending fax hangs up
If your looking into setting up a reliable fax server and your not doing
it over IP, then your best results will be using
Daniel Hazelbaker wrote:
Um..
Your=You're!
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
-- Bandwidth and Colocation Provided by
2009/2/27 John Todd jt...@digium.com:
On Feb 26, 2009, at 6:01 PM, Senad Jordanovic wrote:
Gavin Henry wrote:
Hi all,
In a pure VoIP env, what is the current state of do's and don't s of
virtualizing * in order to provide multiple separate instances, say
for hosting lots of
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 27, 2009 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call file
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 27, 2009 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call file
Agreed, but the OP seemed to be looking for a command-line solution, not a
C one.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 27, 2009 2:20 PM
To: Asterisk Users Mailing
I know the subject of echo has been discussed ad nauseum, but I think I
have a somewhat unusual problem. I am suddenly experiencing occasional
echo on SIP to SIP calls. This is a new development and has never
happened in all the years we've been running *. The phones involved are
not junk
On Fri, 27 Feb 2009, Bruce Komito wrote:
I know the subject of echo has been discussed ad nauseum, but I think I
have a somewhat unusual problem. I am suddenly experiencing occasional
echo on SIP to SIP calls. This is a new development and has never
happened in all the years we've been
It can only be acoustic echo. Asterisk doesn't cancel that - it's the
phone's job.
Maybe it will fix it to reduce volume of the phones.
Steve
On 2/27/09, Bruce Komito bru...@bagel.com wrote:
I know the subject of echo has been discussed ad nauseum, but I think I
have a somewhat unusual
Sent: Friday, February 27, 2009 11:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] call file concurrency
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
On Fri, 27 Feb 2009,
Hello,
If anyone is using a TE121B card and it works reliably (i.e. no HDLC
Bad FCS or similar errors), could you pass along the make, model, and
basic configuration of your Asterisk server?
We tried upgrading our old Dell PowerEdge server to a SuperMicro system,
but that didn't help. I would
Great -- thanks for that.
Brandon.
On Fri, Feb 27, 2009 at 8:34 AM, John Todd jt...@digium.com wrote:
On Feb 26, 2009, at 3:05 PM, Tzafrir Cohen wrote:
On Thu, Feb 26, 2009 at 01:44:19PM -0700, Brandon B. wrote:
At the top of my /etc/dahdi/system.conf file is this line:
#
Any Supermicro server configuration running a mobo with the PDSMA+ chipset has
been bulletproof for us.
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
716-601-4474 MOBILE
I'm working on an Asterisk system with all of it's PRI ports configured with
the LBO setting at 5, like this:
span=3,0,5,esf,b8zs
As of yet, I am unwilling to change the LBO to 0 to where it probably should
be because the system is working and I'm not sure exactly what the LBO does.
I'm
Preparing to use * for a 'real' installation shortly.
Meanwhile, I've got a single port clone thing, 00:06.0 Communication
controller: Motorola Wildcard X100P working to answer my landline and send
calls to my laptop or voicemail. Sweet!
Trying to call out from linphone, I set up this:
On Friday 27 February 2009 14:03:19 Doug Lytle wrote:
Daniel Hazelbaker wrote:
Specifically, I am trying to play around with setting up a fax
server. I can receive the fax, but sometimes the sending fax hangs up
If your looking into setting up a reliable fax server and your not doing
it
Michael Higgins wrote:
exten = _X.,1,Dial(DAHDI/1,${EXTEN})
This should be _X.,1,Dial(DAHDI/g1/${EXTEN})
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Ignacio,
Our Enswitch product matches all these requirements; indeed it goes well
beyond them:
- We scale far beyond 3000-4000 concurrent calls. We'd consider such a
system medium sized. At this size the system is fully
failover/redundant, and we have solved the telephony problems of queues,
I have a couple of suggestions:
Make sure that your timing configuration is correct in
/etc/dahdi/system.conf (that it has a valid timing source).
Also, you probably will probably want to use the half_full buffer
policy, and set the number of buffers used to something reasonable, like
8, to
On Fri, Feb 27, 2009 at 4:14 AM, Christian Victor
christ...@victormedia.de wrote:
2009/2/27 Bill Michaelson b...@cosi.com
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
Afaik only by limiting the number of call files in the
On Feb 27, 2009, at 1:35 PM, Anthony Messina wrote:
On Friday 27 February 2009 14:03:19 Doug Lytle wrote:
Daniel Hazelbaker wrote:
Specifically, I am trying to play around with setting up a fax
server. I can receive the fax, but sometimes the sending fax
hangs up
If your looking into
On Friday 27 February 2009 17:02:16 Daniel Hazelbaker wrote:
Or, if you're using Asterisk 1.6 and looking to try something new,
take a look
at http://messinet.com/AsteriskFAXGateway
I'll take a look at both packages. I hadn't given HylaFAX(+) any
thought as when I searched initially
On Fri, 27 Feb 2009, James Sneeringer wrote:
If you can get the outgoing directory (or a reaonable parent) on its own
mountable partition or volume, you could accomplish this with disk
quotas. It won't control how many Asterisk processes at once (does it
even handle them in parallel?), but
Thank you Gordon and Alexander. With your help, I got it working like
so:
[app-dnd-on]
exten = *78,1,Answer
exten = *78,n,NoOp(${CALLERID(num)} is going on DND ACTIVE)
exten = *78,n,Set(DB(DND/${CALLERID(num)})=On)
exten = *78,n,Playback(do-not-disturbactivated)
exten = *78,n,Hangup
Steve Edwards wrote:
On Fri, 27 Feb 2009, James Sneeringer wrote:
If you can get the outgoing directory (or a reaonable parent) on its own
mountable partition or volume, you could accomplish this with disk
quotas. It won't control how many Asterisk processes at once (does it
even handle
Further to a recent post about a problem whereby the server continues to spew
packets to the phone after hangup (sometimes, not every time), I have found
that this problem appears to be alleviated by using RFC2833 instead of SIP
INFO, however in switching to RFC2833 I introduce another problem
On Fri, 2009-02-27 at 14:07 -0700, Brandon B. wrote:
As of yet, I am unwilling to change the LBO to 0 to where it probably
should be because the system is working and I'm not sure exactly what
the LBO does. I'm aware some changes were made to deal with low audio
levels.
LBO stands for Line
Witness the fact that the old Pingtel phones ran Java, and they were
incredibly lame.
I think part of what this thread misses is that DSP is a god chunk of
what SIP phones need. A general purpose CPU is not the right tool for
the task. A cheap DSP is better suited to compression, transcoding,
Hi all,
thanks for the excellent information about the banks and usb banks.
some tech details will prevent us from using usb units. The trunks will be
500 feet away from the new location of the ip-pbx so we have decided to go
with channel banks for the trunks and sending the E1 signal over cat 5
IIRC, some early dialler of the pre-AMI era used this technique to control
the number of calls placed simoultaneously - they just counted the number of
call files in the spool dir. As they are deleted when the call is over, this
was a simple way to do the throttling.
You could use a similar
On Fri, Feb 27, 2009 at 09:39:59PM -0600, Michael Graves wrote:
Witness the fact that the old Pingtel phones ran Java, and they were
incredibly lame.
I think part of what this thread misses is that DSP is a god chunk of
what SIP phones need. A general purpose CPU is not the right tool for
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