On Sat, Feb 11, 2012 at 8:03 AM, asterisk jobs asteriskcod...@gmail.com wrote:
Hi everyone,
Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about
5000 numbers and then put the call to agents right away and pull up the CRM
based on the number dialed. So, I am going to be
On Wed, Jan 25, 2012 at 10:29 AM, Faraj Khasib fkha...@iconnecths.com wrote:
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is
called but it doesnt execute the command, I tried the command in terminal it
worked, any help please ... below is my dial
On Thu, Jan 26, 2012 at 7:18 PM, David Backeberg dbackeb...@gmail.com wrote:
shebang /path/to/bash
PATH=$1
lame --arguments $1.wav $1.mp3
if [ -f {$1}.mp3 ] ; then
rm {$1}.wav
And my silly code sample hasn't been debugged, and I can spot one
glaring bug, and another less important bug
On Thu, Jan 26, 2012 at 7:36 PM, Steve Edwards
asterisk@sedwards.com wrote:
The OP was using MIXMONITOR_EXEC (although I wonder about the '' syntax)
so he doesn't need to explicitly execute (via system()) his commands.
Wow. Never knew that was possible. I still don't like the syntax, but
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie
m...@chrishowie.com wrote:
I've been up and down this issue for a few hours and I cannot for the
life of me determine why simply defining a peer causes Asterisk to offer
telephone-event. I have tried specifying dtmfmode=rfc2833 or
On Thu, Jan 5, 2012 at 8:05 AM, Steve Underwood ste...@coppice.org wrote:
No PAP2 or PAP2T supports T.38, even though many people will swear that they
do. For a little while there was some beta code for the PAP2T with badly
broken T.38 support. Perhaps this is where the legend of T.38 on a
On Wed, Jan 4, 2012 at 4:45 PM, Asterisk Development Team
asteriskt...@digium.com wrote:
The Asterisk Development Team is pleased to announce the first
release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.
2.6.0 is a feature release which:
wct4xxp: Expose serial number in dahdi_device and
On Wed, Dec 28, 2011 at 4:10 PM, Danny Nicholas da...@debsinc.com wrote:
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
What's the difference between a car released in 2006 versus a car
On Fri, Nov 18, 2011 at 2:23 PM, Sazzad sazzadbinka...@gmail.com wrote:
Hi,
I have to use asterisk with some dedicated DSP chips, which will do the
expensive G729 CODEC computing, so that the server processor has minimum
load. I was informed, I've to use GPAK to implement this. So far I've
I
On Thu, Nov 10, 2011 at 12:24 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 11-11-10 12:12 PM, Danny Nicholas wrote:
Yeah! My boss will be much happier having a system that doesn't have the
-tail on it.
I hear this kind of statement every once in a while, which makes absolutely no
On Thu, Oct 27, 2011 at 11:53 AM, Mike l...@net-wall.com wrote:
I am trying to record a MeetMe conference, and this is what is relevant in
the 1.8 manual:
r - Record conference (records as MEETME_RECORDINGFILE using format
MEETME_RECORDINGFORMAT. Default filename is
On Sun, Oct 23, 2011 at 3:16 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Wed, Oct 19, 2011 at 10:11:08AM -0400, David Backeberg wrote:
If you use DAHDI, you need to change ownership of /dev/dahdi/* to the
non-root owner. I ended up rolling that into the init script for
dahdi
On Wed, Oct 19, 2011 at 7:19 AM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
Thank you, I actually found the asterisk.conf settings after sending the
mail. So next question is which folders/files do I need to change ownership
of to make it work?
/etc/asterisk
On Mon, Sep 12, 2011 at 11:19 AM, Tarek Sawah tareksa...@hotmail.com wrote:
i wanted to upgrade to 1.6 .. i did and when tested it .. the server crashed
at 100 concurrent calls.
please advise?
Nobody will know why your asterisk crashed unless you follow the
instructions here:
On Tue, Sep 6, 2011 at 10:36 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
However you could select/deselect modules using menuselect if you wanted to
automate the process. It's documented over here:
http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439
I'm having annoying errors trying to get configure working.
tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz
cd asterisk-1.8.6.0
./configure
I get complaints related to pwlib / ptlib...
checking for openr2_chan_new in -lopenr2... no
checking /root/pwlib/include/ptlib.h usability... no
checking
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Fleming kpflem...@digium.com wrote:
This is a bug in the configure script, but in the meantime, you should be
able to use --without-pwlib to avoid it, as long as you aren't trying to
build chan_h323.
Thanks much.
I was trying
./configure
read the 1.6 README and the 1.8 README.
If you're using SIP you should expect changes with account
authentication, faxing, output regarding channel status and
performance.
I think that version of 1.4 is late enough you would already be on
DAHDI for hardware devices. If not, you need to convert
That debug looks cool but I have no idea what it means.
If you are using T.38, turn it off, and do audio fax, recorded with MixMonitor.
When you can hear the audio of the fax hopefully you will be able to
tell what's going on, and if you're lucky it's something specific to
the particular kind of
On Mon, Jul 11, 2011 at 5:29 PM, Steve Edwards
asterisk@sedwards.com wrote:
Many times, I've made the statement that you can execute hundreds of AGIs
written in C in the time it takes to load an interpreter and parse a script
written in PHP or Perl.
I've truly enjoyed this thread. And
On Mon, Jun 27, 2011 at 9:06 AM, Michael voip.quest...@gmail.com wrote:
Hi Kevin,
Controlling it through the sip.conf peers is sufficient for us for this case
(because this particular provider doesn't support T.38 at all), but I think
it would be a good idea to add the option to
On Thu, May 5, 2011 at 1:43 PM, vip killa vipki...@gmail.com wrote:
The majority of open source projects out are NOT run by commercial
institutions...
Postfix kicks butt. But only because IBM paid for development, for a
long number of years, and because they hired somebody who had a really
good
On Wed, May 4, 2011 at 12:00 PM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
(For my part, I'm actually surprised that nobody came up with a proper
protocol for encapsulating the stream of zeros and ones that make up a fax
transmission but rely on the precise timing inherent with a
On Mon, Apr 25, 2011 at 10:40 AM, C. Savinovich
c.savinov...@itntelecom.com wrote:
Does this ConfBridge requires a hardware timing source?
No, and neither does MeetMe with modern DAHDI.
Will I be able to use this on any virtual server without having the need
special changes to
the VM
On Fri, Apr 1, 2011 at 7:04 AM, Khaled W. Chehab kche...@xplorium.com wrote:
1-Is there a way to export fax tiff file image from .pcap captured file .
Maybe, but I can't think of how. If you can somehow invert the pcap
file back into packets and reproduce the fax traffic, then maybe.
In other
2011/2/3 Marcello Colucci (SIRIO Informatica s.a.s.)
mcolu...@sirioinformatica.it:
Hi, I have asterisk 1.6.2.6 on a Debian Lenny system.
When I try to send a fax in T.38 mode I receive this error
ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel
'SIP/eutelia-sirio-out-' is in an
On Tue, Jan 25, 2011 at 7:01 PM, Bryant Zimmerman brya...@zktech.com wrote:
Ok If I set t38pt_udptl = no on the trunk the fax comes in t.30 but I can't
make t.38 work I keep getting the following error Disconnected after
permitted retries Any ideas on this?
So you're saying if you turn off
On Tue, Jan 25, 2011 at 9:34 AM, Bryant Zimmerman brya...@zktech.com wrote:
On 01/24/2011 2:54PM Bryant Zimmerman wrote
The attached file was too large so I am putting in a link to the file. It is
a virus free text file.
You failed to mention earlier that this is T.38.
Turn off T.38 and see
On Tue, Jan 25, 2011 at 1:45 PM, Bryant Zimmerman brya...@zktech.com wrote:
Do you know how to force off T.38 in res_fax?
it's in sip.conf
take a look for
t38pt_udptl=yes
change it to no
reload sip
on your console
that should force it to either fail entirely or do audio passthrough.
--
On Mon, Jan 24, 2011 at 2:53 PM, Bryant Zimmerman brya...@zktech.com wrote:
I am testing out inbound faxing using res_fax and res_fax_spandsp.so
My system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries.
How would
On Mon, Jan 24, 2011 at 4:51 PM, Steve Edwards
asterisk@sedwards.com wrote:
We know the problem exists -- the boss just installed U-verse at his house
:)
It works fine from cell and copper, just not from U-verse and their ilk.
Well, I would say more data samples are needed then. It could
On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepal ami...@phoenixinternet.net wrote:
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can
send recieve faxes from both boxes fine to and from pstn. But the faxing
between 1.6 and 1.4 extensions does fail. Any ideas please ?
You
On Wed, Jan 5, 2011 at 6:59 PM, Myles Wakeham my...@techsol.org wrote:
For some reason our Asterisk box is doing something really unusual following
applying a routine update to CentOS 5 on Monday.
We have Asterisk 1.4.2 and its been working great for years. But now when
the phone system
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com wrote:
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started asterisk still no CDR's
flowing to
On Wed, Dec 8, 2010 at 9:06 AM, Gilles codecompl...@free.fr wrote:
Hello
I need to find a recent and neutral comparison of the major products
available to connect an Asterisk server to the telephone network,
whether ISDN (BRI) or PSTN, and through a PCI card or some external
box. I'm
On Wed, Dec 8, 2010 at 10:17 AM, Gilles codecompl...@free.fr wrote:
On Wed, 8 Dec 2010 09:33:22 -0500, David Backeberg
dbackeb...@gmail.com wrote:
* pay somebody else to do it in the form of appliance and lose most
control versus do it yourself and have total control but also the
chance to screw
On Tue, Nov 23, 2010 at 8:25 AM, voip crazy voipcr...@gmail.com wrote:
Hello,
I want to analyze the asterisk logs files, looking for all kind of
errors, ¿Anyboby knows any asterisk logs analyzer?
You're only going to have the logs for what you create logs for.
I create custom logs for the
On Tue, Nov 30, 2010 at 7:34 PM, Duane Larson duane.lar...@gmail.com wrote:
I have MySQL Cluster set up for OpenSIPS which allows for the best Redundant
High-Availability. I was wondering if it's possible for Asterisk to also
use multiple database servers for Realtime? Currently with Realtime
On Sun, Nov 28, 2010 at 5:26 PM, dotnetdub dotnet...@gmail.com wrote:
Sorry,
what I meant was:
server*CLI remove extension (hit tab)
segfault..
1.4.22
It could be an extension name Where is the error trapping if this is the
case.. Who writes this shit?
If you remove an extension that
On Mon, Nov 22, 2010 at 8:47 AM, Vilius Adamkavicius
vilius.adamkavic...@invade.net wrote:
Hi All,
We have a requirement to record over 60 simultaneous calls. Our recording
facilities are implemented using Monitor() over AMI. The thing we have
noticed that making 60 simultaneous call
On Mon, Nov 15, 2010 at 8:30 AM, Richard Kenner ken...@gnat.com wrote:
It's kind of low for me. How does one control that volume?
I've never heard of a way to control that volume.
You can tweak after-the-fact with sox, or you can crank up your
soundcard / amplification on playback.
--
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch ca...@usawide.net wrote:
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far away, and process the calls?
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question voip.quest...@gmail.com wrote:
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote:
Hello,
I'm trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn't recognize that we have this capability and sends a 488
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote:
It's set to yes for this peer.
also t38pt_udptl is set to yes.
:(
You don't say anything about what you're trying to send / receive against.
Here's how you should troubleshoot:
* start with a 'real fax machine'
On Tue, Oct 19, 2010 at 10:23 AM, marvin horst fivehor...@gmail.com wrote:
How did the setup work as far as extensions on the Inter-Tel system
contacting extensions on the asterisk system?
It worked, I dare say, flawlessly. Well, as flawlessly as Inter-Tel
worked. Still had to watch out for
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question voip.quest...@gmail.com wrote:
The whole point (as I specified in the header and initial message) is the
attempt to use Fax for Asterisk to send the message.
Asterisk can handle audio passthrough faxing. I'm talking audio faxing
over SIP. You
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question voip.quest...@gmail.com wrote:
Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll dan...@danielknoll.de wrote:
Hey,
i forgot to ask, how can i get the user number from a caller he is in a
conference, i don't find a variable to us this for the current channel.
Only the command meetme list roomnr shows the usernumber, but i can't
On Wed, Oct 6, 2010 at 5:00 PM, marvin horst fivehor...@gmail.com wrote:
Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone
system via a SIP trunk using the IPRC card?
I have, believe it or not, integrated Asterisk with Inter-Tel.
However, not via SIP. Run the costs.
On Thu, Sep 30, 2010 at 10:51 AM, khalid touati khalidtou...@gmail.com wrote:
Hi List,
I did follow the procedure to install Free Fax for Asterisk successfully
till i came accross this isssue: i can't load the fax module:
pbx3*CLI module load res_fax_digium.so
Unable to load module
On Thu, Sep 30, 2010 at 11:46 AM, khalid touati khalidtou...@gmail.com wrote:
thanks for replies,
I am using Asterisk 1.6.2.11
and components res_fax-1.4_1.2.1-x86_64 and
res_fax_digium-1.4_1.2.1-barcelona_64.
(amd 64 bit machine)
actually I am not aware that there is version which include
On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
The reason is when doing a load balancing , We cannot confine the
recording to a particular asterisk machine ( If we have more than one
asterisk machine in the topology ).
Yes you can. You can
On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote:
Don sez: I don't know how to make Outlook indent. I usually top-post, but I
don't like getting yelled at.
Why do you say Don't do that? Is there a real reason that it would be bad?
Performance is a real reason. Multiple
On Thu, Sep 23, 2010 at 2:21 AM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
HI ,
Is there Any way is there so that I can store my recordings directly to a
database rather storing the same to a file .
Please, please, please tell us why you would want to do that.
--
On Wed, Sep 22, 2010 at 10:00 AM, Adam Moffett a...@plexicomm.net wrote:
In the simplest terms I can think of, I'm going to describe what I want to
do and I want to know if it's possible in the current version of asterisk.
Can I take a T38 call from an ATA, convert that back to analog and have
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Now I have no previous experience with Cisco systems and don't want to screw
up anything. Are they much different than Asterisk based systems? I guess
the underlying VoIP technology is the same for both the systems so it
On Tue, Sep 14, 2010 at 3:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Now I have no previous experience with Cisco systems and don't want to screw
up anything. Are they much different than Asterisk based systems?
sometimes. Cisco supports SIP, but depending on the product,
asterisk
On Mon, Sep 13, 2010 at 4:33 PM, Stanislav Korsei kor...@rinogo.com wrote:
Hello!
I've created clean installation of Asterisk 1.6.2.11 with spandsp 0.0.5.
When i try to receive fax I get:
[Sep 13 00:45:59] WARNING[3283]: app_fax.c:432 transmit_audio: channel
'SIP/crocus-ua-0004' refused
On Wed, Sep 8, 2010 at 4:18 PM, Stanislav Korsei kor...@rinogo.com wrote:
Can you recommend any specific solution to this problem or way to install
app_fax?
Not without specific debugging about what problems you're seeing. You
get a lot of information when faxes succeed or fail. Try a fax and
On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com wrote:
The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk
SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
PSTN.
You don't say the percentage that are failing. However, people
On Tue, Aug 24, 2010 at 9:05 AM, Ron nha...@gmail.com wrote:
hi all,
i recently subscribe for an isdn and terminate it on a 3825 router.
i used it as a sip trunk for my asterisk. i'm a newbie when it comes to
ISDN. and i've been experiencing some issues:
1. Call Hangup:
When hangup is
On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull dun...@e-simple.co.nz wrote:
Voice recognition is a pain for people with accents and poor lines and when
Everybody has an accent. Some people live in a place where the people
they talk to sound like themselves, so they forget that fact.
Of
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote:
We gave the phone a static IP address and pointed it to the configuration
server on the remote end that has the CFG files for it. The phone starts
up, downloads SIP and the “new application” and otherwise seems to be
On Fri, Aug 13, 2010 at 11:43 AM, Eric Merkel (Mail Lists)
ejmerkel.li...@gmail.com wrote:
I am looking to build a small PBX for an office that has 3 incoming analog
lines and less than 10 extensions.
For that small of an installation you might prefer an asterisk
appliance. You can review the
2010/8/13 Lyle McKarns lyle.mcka...@nexusmgmt.com:
Does anyone have any feelings one way or the other about running Asterisk on
AMD vs running Asterisk on Intel?
Only political feelings. I want to support AMD so there's at least
some token competition for Intel.
Both companies make nice 6-core
On Wed, Aug 11, 2010 at 10:12 AM, Kent Varmedal k...@domeneshop.no wrote:
I'm trying to set up an old PBX (that supports SIP) to go through our
new Asterisk server, so that our old phones can be used still for some
time.
How can I set up Asterisk to deliver a trunk sip connection that our old
On Wed, Aug 11, 2010 at 4:36 AM, Tino t...@sparksupport.com wrote:
Is it possible to install Asterisk on Vmware(centos) from source. Is there
any difference or disadvantage for this compared to asterisk running on
physical machine.
This has come up repeatedly on the list.
Basically, the less
On Wed, Aug 11, 2010 at 11:24 AM, Kent Varmedal k...@domeneshop.no wrote:
We need to upgrade this PBX for it to work with SIP, it is at the moment
using ISDN. And those who delivered it and do the
support/reconfiguration is paid by the hour. We don't have any control
over it our self, so when
On Mon, Aug 9, 2010 at 4:36 AM, Zhang Shukun bit...@gmail.com wrote:
hi, group
there are two module can used for meeting. MeetMe and
Conference(which is a plugin)
My question is :
which is better for large conference(maybe above 100 people in a meeting)?
There's at least one more
On Wed, Aug 4, 2010 at 11:57 AM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Not really an asterisk
On Thu, Jul 29, 2010 at 4:15 AM, zeynep yildirim zyildi...@gmail.com wrote:
Hi All,
I upgraded 7970 from SCCP to SIP. But the phone isn't registering.
Have you got any working XML file for 7970 phones.
Isn't registering with what?
If you're registering that with CallManager, you have to
On Thu, Jul 29, 2010 at 7:22 AM, Nguyen Quang Tri kihote...@gmail.com wrote:
Hello,
i have Cisco Unified Communications Manager with 10 ip phone,i dont buy
license IVR of Cisco Unified Communications Manager. Can i use feature IVR
on Asterisk connect with Cisco Unified Communications Manager.
So historically I've done one of two things on systems where I've
needed to use MeetMe
* used a real Digium card, and I've only ever used a TE400 or a TE420
for that purpose, and I know they have the timing chip
* used dahdi_dummy, which works well with light load, but I had it
running on a very
On Thu, Jul 29, 2010 at 5:04 PM, unsero...@aol.com wrote:
Do you know if it is possible to interconnect 1.6 with Microsoft Office
Communications Server 2007 and use the Office
Communicator as a softclient for telephone calls and the Communicator for
Instant Messaging? I believe you can set up
On Tue, Jul 27, 2010 at 6:08 PM, bruce bruce bruceb...@gmail.com wrote:
:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
not good for Queues at all. It's using the reporting tool from Areski and
One of the several things you asked for was GUI for cdr database logs.
On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote:
I seem to not be able to find any good open source Asterisk Queue Analyzer
and Asterisk Log Analyzer on the web.
google 'freepbx'
It does some of what you want. For the rest of what you want, strongly
consider paying a
On Fri, Jul 23, 2010 at 8:46 AM, Matt mhop...@gmail.com wrote:
It's not necessarily this simple. There is an approximately 50-75foot cable
run through ceilings and walls (CAT5) to the location where the phones will
be. At the phone location there is no power.
You always have options. You
On Thu, Jul 22, 2010 at 2:46 PM, Matt mhop...@gmail.com wrote:
I've got an interesting situation where I have one cable run from the feed
area to the service area. I have three devices that I need to power at the
service area. Is anyone aware of a device that will take the POE from the
On Sat, Jul 17, 2010 at 6:52 PM, David Shauger sollost...@gmail.com wrote:
Can anyone provide the settings in Audacity to create a proper wav file
without having to do additional conversion in the cli? Has to be a way to do
this with less steps.
If your goal is to 'minimize steps', you should
On Sat, Jun 26, 2010 at 2:09 PM, Eyal Goltzman egoltz...@gmail.com wrote:
Hello,
After installing and learning Asterisk I found myself with a need for a
minimal set of empty configuration files with only the must have stuff in
order to setup a SIP only machine, is there a place to find it?
On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch ca...@usawide.net wrote:
But, we have an opportunity to get into a big time telecom activity.
It would have 2000 to 30,000 user lines per city, and we would like to have
those brought back to a central location for control and because transport
can
On Fri, Jun 25, 2010 at 11:00 AM, Cary Fitch ca...@usawide.net wrote:
I see some talking about TNTs in this forum. Those are 672 lines or in some
versions double that, what is used behind them to do the processing, etc.
So a channelized DS3 is roughly 28*23 channels in US if you do one
On Mon, Jun 21, 2010 at 3:04 PM, Necati Demir nde...@demir.web.tr wrote:
This is a really rookie question: when should i use TE110P ISDN PRI Card?
--
Necati DEMİR
When you have a single PRI / BRI line you wish to terminate into an
asterisk system.
--
Hello List:
I'm working on a funny scenario, where I'm bouncing calls from a Cisco
call center into asterisk. Cisco call center has some logic that if a
customer calls in, an agent is logged into a given extension... if
Cisco sends a customer call to that extension, and there is a ring
with no
On Wed, Jun 16, 2010 at 11:50 AM, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 16 June 2010 08:21:17 David Backeberg wrote:
I know if I do not do an Answer() that the call is not yet picked up.
However, if I do a HangUp(), is that functionally equivalent? Can you
Hangup() a channel
On Sun, Jun 13, 2010 at 2:59 PM, Vieri rentor...@yahoo.com wrote:
I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
compatible).
I've taken a look at CAGI and QUIVR but their latest code releases date back
to 2006.
I've also seen a more recent project (wildpbx)
On Thu, Jun 3, 2010 at 3:56 PM, cov...@ccs.covici.com wrote:
Hi. For several months now asterisk will mysteriously stop inserting
records into cdr database. I am using mysql and the asterisk addons
1.6.2 to accomplish this. Sometimes there is a strange error about
column names, but often
On Thu, May 27, 2010 at 6:17 PM, Theo Band theo.b...@greenpeak.com wrote:
I used to build Asterisk from source including the zaptel-dummy module.
Last year I decided to upgrade and use a yum repository. I hoped that
this would be less hassle compared to manually chasing after the latest
On Mon, May 24, 2010 at 9:41 AM, Kingsley Tart kings...@skymarket.co.uk wrote:
I know nothing of Trixbox but I had a problem with my own dialplan where
there was a delay with the user selecting 0 from my IVR menu. It turned
out that because my extensions all started with 0 (they were real phone
On Thu, May 27, 2010 at 4:05 AM, Theo Band theo.b...@greenpeak.com wrote:
First I noted that dahdi_dummy is no longer present in
kmod-dahdi-linux-2.3.0.1-1.
Not exactly true.
myhost01 asterisk # lsmod | grep dahdi
dahdi_dummy 5812 0
dahdi_transcode 8968 1 wctc4xxp
On Mon, May 24, 2010 at 7:31 AM, Marcin J. Kowalczyk
marcin.kowalc...@ccig.pl wrote:
Medium load system (~300 simultaneous calls) crases few times a day.
1.6.1.19 but then upgraded to 1.6.2.7 but it's not solving issue.
Any idea what can be wrong/tunned?
I've three times had unexplained
On Thu, May 20, 2010 at 11:41 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm evaluating what could keep me from upgrading production systems to
1.6.2.
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an
issue with BLF-pickup which kept me from going further.
Have you met
On Wed, May 19, 2010 at 12:38 AM, Adolphe Cher-Aime achera...@gmail.com wrote:
Hello Everyone,
I must deploy an asterisk system that can support
at least 500 pstn outbound calls.
It's a challenge as it's the first time i'm gonna build such a large
system.
I want to
On Tue, May 18, 2010 at 6:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I read on voip-info.org that Asterisk 1.4 support T38 passthrough.
That may or may not be true. I do not know.
I do know that I've had much better success with fax in 1.6 than I
ever had in 1.4.
My personal
On Wed, May 12, 2010 at 7:45 AM, Steve Underwood ste...@coppice.org wrote:
On 05/12/2010 08:46 AM, David Backeberg wrote:
So buy an asterisk appliance that supports fax, and then you can pay
somebody else to do the upgrade.
Does that appliance actually support FAX? The web pages don't mention
On Wed, May 12, 2010 at 9:53 PM, Ben Dinnerville b...@voicelogic.com.au wrote:
We are still getting an issue with a particular file which I have tried
multiple different ways to create to no avail. The tiff file is created
with ghostscript from a pdf as per the guidlines but every time we try
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com wrote:
This one works on my box (Vestec on 1.4.30 on OpenSuse)
Hmm... Not for me.
$Digit = (ONE:1 |
TWO:2 |
THREE:3
Ummm, zed is z.
I was thinking of nought.
On Tue, May 11, 2010 at 8:39 PM, David Backeberg dbackeb...@gmail.com wrote:
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard Kenner ken...@gnat.com
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a week), but it appears they no longer offer their product
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