Rizwan Hisham wrote:
Well database really is a bottleneck for me. I am currently trying to do
rating stuff in agi using perl. What im doing is i lookup the rate of
every dialed code for every call from the mysql database using the
longest match technique. The longest match technique costs
SIP wrote:
Tilghman Lesher wrote:
We're about to do another batch of sounds, and I see by my word count that we
have some extra time left over. So, suggestions will be entertained for
additional prompts in English, Spanish, or French. The only rules are: 1)
the
prompts have to be generic
Andreas van dem Helge wrote:
Does anyone have a script for manual wardialer for asterisk? not sure
if wardialer is the correct term but basically I want to call X
number say 555- through 555-0050 and be able to listen to each
call and when I hang up or press a key it will dial the next
Andres wrote:
We have tested both and they work fine. The Sangoma is much easier to
install as it does not depend on any other driver, you just run
'setup-sangoma' and follow the instructions. You don't have to fiddle
with the linux kernel or zaptel or chan_misdn. It just works. Plus
I'm having a problem at a custom site where GotoIfTime doesn't seem to be
working for some reason. I had putty running and logging CLI output and below
is the call data:
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Ringing(Zap/3-1, ) in new stack
-- Executing Wait(Zap/3-1, 0) in
Lee Jenkins wrote:
I'm having a problem at a custom site where GotoIfTime doesn't seem to be
working for some reason. I had putty running and logging CLI output and
below
is the call data:
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Ringing(Zap/3-1, ) in new stack
Brent Davidson wrote:
John Signorello wrote:
excuse me...
But did you not just post
[asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New
Digium Boards Cheap X305 $199
Did you not provide a link to a COMMERICAL entity?
Wasn't your a post a unsolicited post, that is, not in
Al lists wrote:
Hi list,
Any good drag and drop transfer call application for windows based
systems you can advise ?
Something like HUD perhaps?
Yes.
Maestro Control Panel (I authored this one)
http://www.datatrakpos.com/pos/datatalk/maestro.aspx.
There is also the nice flash based
Lee Jenkins wrote:
Al lists wrote:
Hi list,
Any good drag and drop transfer call application for windows based
systems you can advise ?
Something like HUD perhaps?
Yes.
Maestro Control Panel (I authored this one)
http://www.datatrakpos.com/pos/datatalk/maestro.aspx.
There is also
Bob G wrote:
Introducing Click-to-Call http://1ezphone.com/
Posted: 16 Apr 2008 9:55 AM PDT
The 1EZphone browser softphone has created so much buzz in the media
that a lot of individual users and companies who have a web-presence;
Websites, Online Advertising, Blogs, Customer support
Steve Edwards wrote:
On Wed, 16 Apr 2008, Bob G wrote:
Why the guy asked a question?
From: Lee Jenkins
Bob G wrote:
Introducing Click-to-Call
I think you're going to get yelled at ;)
1) You hijacked the thread.
2) You top-posted.
3) It's a non-commercial list
Kai-Uwe Jensen wrote:
An app to invoke the Cepstral text-to-speech engine.
On Tue, Apr 15, 2008 at 11:46 AM, Zoa [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
What is app_swift ?
Zoa
I've written an AGI wrapper for it as well, in case you don't want to
re-compile
to
Darryl Dunkin wrote:
FOP works for us, no need for X:
http://www.asternic.org
If you need to avoid using a mouse, you can use the Polycom attendant
console instead:
http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound
point_ip_attendant_console.html
We recently
and selecting my record :) == Lee Jenkins.
Are you having trouble entering the software? If so, it's a good chance you
may
be running Vista or a nicely locked down version of XP. The original installer
saved Maestro's (firebirdsql) database file to the (\program files\Maestro
Control Panel
Brent Davidson wrote:
I'm having a major problem at one of my branch offices with Phantom
Rings on their asterisk-based phone system. The system was originally
built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC
card. The upgrade severely increased the frequency of the
Vincent wrote:
On Thu, 3 Apr 2008 10:51:15 -0430, Earl Terwilliger [EMAIL PROTECTED]
wrote:
http://www.micpc.com/eventmonitor/
Thanks guys. I was also thinking of stand-alone apps like Jabber or
something. The call is simply to know if an extension is on- or
offline.
Not web
Announcement:
We are pleased to announce that we have released AsterPas FastAGI ObjectPascal
Script Server for Asterisk PBX under license.
What is AsterPas?
AsterPas is a FastAGI server which allows real-time scripting of Asterisk PBX
call flow using ObjectPascal based scripting.
Lee Jenkins wrote:
Announcement:
We are pleased to announce that we have released AsterPas FastAGI ObjectPascal
Script Server for Asterisk PBX under license.
Oops. That should be LGPL license ;)
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling
Tong wrote:
That might work.
I'll give that a shot.
Doug Lytle [EMAIL PROTECTED] wrote:
Tong wrote:
Is there a way to limit outbound calls when feeding files to the outgoing
directory in asterisk? I several thousand files i need to feed asterisk,
hoping to copy it to the
Jerry Geis wrote:
Jerry Geis wrote:
I am running asterisk 1.4.18 trying to use MeetMe and option b.
I am getting permissions denied failed to execute conf-background.agi
on the CLI
lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -
/home/silentm/bin/conf-background.agi
my
James Finstrom wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Anyone have the telemarketer torture prompts? I would seriously like
to revive this.
- --
I wrote one a while back that uses Cepstral TTS, but the mechanics are simple.
When a telemarketer calls, I say hmmm, that
[EMAIL PROTECTED] wrote:
Is there way to get the logs of the call generated by Manager API, or is
there some other way to achieve same scenario so that I can get the status of
the call generated by me.
Actually I have a scenario where I have to call customers and play a message,
I do
[EMAIL PROTECTED] wrote:
I am generating an outbound call through the Manager API and bridging it to
an internal Extension, my problem is I am not able to find the logs for the
call generated by the Manger API, Since on the same Asterisk server there are
many users connected and I am
Mark Hamilton wrote:
I don't think the link that Lee gave works.
Also, I wrote a Windows based utility for viewing AMI packets and testing
AMI
commands. It's Freeware:
http://www.voip-info.org/wiki-Asterisk+GUI#OperatorManagerInterfaces
Look for Manager API Test Utility
or
Mark Hamilton wrote:
I don't think the link that Lee gave works.
Oh boy, you were talking about the link to download the software and I
completely misunderstood. My mistake, the link is fixed to download the
software.
Here's the direct link:
John Meksavan wrote:
Asterisk Users,
I am running Asterisk-1.4.11 on a Debian Etch system. On an
occasion, when customer calls into my Asterisk Box, I get this error
messagefrom Asterisk CallerID returned with error on channel Zap/3-1 ,
causing all my zap channels to be busy. So, I
Raul Alarcon wrote:
im trying to use monitor() aplication with b option, to start the
recordigin just once the conversation has actuallly begun.
It works fine with a sip extensión, but when i use a zap channel, it
records all the channel bridging, including the ringing sounds...
could
Chris Bagnall wrote:
Greetings list,
I've been playing around this afternoon with Flash Operator Panel, trying to
get it to do attended transfers. I am running the latest version.
Has anyone managed to get this working reliably, and if so, would you mind
sharing how you did it please?
Mitchell Jackson wrote:
Hello! I am having problems figuring out how to do something, and any
help would be much appreciated.
I would like to use the manager API to take an existing call on a
specific SIP extension, dial and conference in a third party.
From what I can tell, the way
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is possible to
do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a call.
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is
possible to do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number
Bill Andersen wrote:
Has anyone tried to used VB6 to communicate with the Asterisk Manager?
If so, would you be willing to share some basic code showing your
approach to getting connected and parsing results?
I've got a Telnet control that is allowing me to connect, authenticate
and see
Soumya Kat wrote:
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and
Stefan Reuter wrote:
Lee Jenkins wrote:
I thought that the OP was asking for something to perl what Asterisk-Java
does
for java coders. I would definitely consider Asterisk-Java to be a
framework,
though not so much with PasAGI which is more of an class object wrapper
around
AGI
Moises Silva wrote:
Asterisk 1.6 includes a new feature that allows using AMI as a transport
for AGI commands, there abstraction becomes even more important.
For Asterisk-Java I am currently adding support for that in a way that
allows the developer to run the same AGI code either through
Alex Balashov wrote:
Well, no, there really aren't any prebuilt high-level frameworks for
approaching Asterisk through the Manager API or AGI. Instead, there are
just AGI bindings that allow you to integrate dial plan logic with
outboard code.
I thought that the OP was asking for
Lee Jenkins wrote:
Jared Smith wrote:
On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
recall.
Thanks
Alex Balashov wrote:
Ken D'Ambrosio wrote:
Hi, all. I've used the perl/AGI interface, and... well, I found it kind
of hokey. Granted, this was in 1.2 days -- perhaps things have changed.
Regardless, I guess I have two questions:
1) Has the Perl/AGI binding improved since then?
2) Is
Hi all,
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Thanks !
--
Warm Regards,
Lee
Everything I needed to learn in life, I learned selling encyclopedias door to
door.
___
-- Bandwidth and
Jared Smith wrote:
On Fri, 2008-02-01 at 15:32 -0500, Lee Jenkins wrote:
What format is the LastCall variable of QueueMember event? I'm looking at:
1201897536 for instance.
Unix epoch format, or number of seconds since Jan 1, 1970 UTC, as I
recall.
Thanks.
--
Warm Regards,
Lee
Vincent wrote:
Hello
I'm pretty much a newbie when it comes to C, but I have to use
this language to write a couple of AGI proggies because I need them to
be statically compiled.
Strangely enough, Google didn't return much when looking for the
Hello, world! of AGI in C.
The
Vincent wrote:
On Sun, 27 Jan 2008 09:09:59 -0500, Lee Jenkins [EMAIL PROTECTED]
wrote:
Sorry, I don't have a sample for you as I write mostly in Freepascal/Lazarus
these days and use my own library for AGI/FastAGI. That said, did you try
saving the file to a fully qualified path?
My
Doug Lytle wrote:
Michael Munger wrote:
Polycom Provisioning Tool Updated.
Looks like a Windows only tool. Shame it doesn't work under Wine.
Doug
Looks like it was written with VB.net. Not sure where Mono is as far as VB.net
goes, but if I'm not mistaken, once its compile it
Lee Jenkins wrote:
Doug Lytle wrote:
Michael Munger wrote:
Polycom Provisioning Tool Updated.
Looks like a Windows only tool. Shame it doesn't work under Wine.
Doug
Looks like it was written with VB.net. Not sure where Mono is as far as
VB.net
goes, but if I'm not mistaken
After having misunderstood some key elements of AstManProxy, I started to write
my own proxy server for Asterisk AMI. I was under the impression that it
required a mysql database to cache its data for some reason. (Is there another
AMI proxy that uses a mysql database?) At any rate, I had
Julian Lyndon-Smith wrote:
Lee Jenkins wrote:
Julian Lyndon-Smith wrote:
astmanproxy does this already, I think ..
Julian.
Of course ;) AstManProxy is a great product from what I had read up on it.
One thing is that it requires (if I'm not mistaken) an mysql installation
which
is too
Hi all,
I'm writing a real-time (not RealTime) proxy server for the AMI interface.
Although I'll be using it for some commercial products, the proxy software
itself will be released under GPL.
I was wondering if there would be any interest in testing it from the
community?
I don't have
Julian Lyndon-Smith wrote:
astmanproxy does this already, I think ..
Julian.
Of course ;) AstManProxy is a great product from what I had read up on it.
One thing is that it requires (if I'm not mistaken) an mysql installation which
is too heavy of a dependency for some applications that
Philipp Kempgen wrote:
Bhrugu Mehta wrote:
I am new to zaptel programming.
can anybody help me how to start this. or any ref. site or matirial
availabel.
i want to use c lang. for this.
sarcasm mode=SCNR class=ignore
Some tutorials:
http://www.google.com/search?q=learn+c+in+21+days
Vincent wrote:
On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins [EMAIL PROTECTED]
wrote:
I have to reboot my desktop xp box daily for it to run well.
I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a
bunch of apps open at all times. And this is a 300E no-name box
srinivas Antarvedi wrote:
Hello users,
Actually i wanted to implement Text-To-Speech engine
from cepstral voice using swift application
i tried the documentation of doing this and i was unsuccessful
at doing this work with asterisk
can anybody please help me out finding the solution to
srinivas Antarvedi wrote:
Hello all,
Here is the requirement from my side
to use Asterisk.NET API to generate
an automated call (outgoing) from asterisk
and then link to one of the extensions which
plays a sound file for the callee.
For this i have worked out in the follwing way
Doug wrote:
At 19:55 12/13/2007, Vincent wrote:
Hello
I was wondering why there doesn't seem to a Windows version of Zaptel,
making the Digium and its clones unavailable for a Windows PBX.
Is the Zaptel/Zapata combo too *nix-centric?
Thanks.
Windows is a half-baked, dying OS
D4rk F1ber wrote:
So I have my asterisk box up and working internally at home and all is
good so far. The next thing I wanted to do was make and recieve calls
to regular land lines now.
I don't have a POTS line and was looking for probably a SIP trunk.
I have seen mentions of Skype
Steve Edwards wrote:
On Sat, 13 Oct 2007, Lee Jenkins wrote:
I have been using axVoice.com for some about 9 month to a year now and
their service is pretty damn good. For home users they have unlimited
plan for around 22.00-24.00 U.S. per month.
I think the pay as you go plans make more
Pirlouwi wrote:
Hello,
I wonder if there is a way to build my own asterisk application (let us
say apps/app_myappl.c),
and to launch other existing applications from it (for example, doing an
apps/app_dial.c, or others).
Could someone highlight me on that?
thx
Pirlouwi.
Even better
Mojo with Horan Company, LLC wrote:
No, because then asterisk would be presented three arguments: '-rx',
'extensions', and 'reload' -- as 'extensions' is not a command by
itself, and the 'reload' appears superfluous to asterisk, this would not
work as desired.
Asterisk needs to be
Yehavi Bourvine +972-8-9489444 wrote:
Hello,
I see that most people are using the extensions.conf syntax (most of the
examples and questions here use that syntax). recently I've translated all my
dial plan to AEL syntax and I find it much easier, especially when you need
IFs.
Why
Michael Munger wrote:
I am trying to use PHP to reload the extensions in an Asterisk
installation. I keep getting this error:
Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
when I run the script by visiting the URL; however, if I run the script
from the command
Steve Totaro wrote:
David Gomillion wrote:
On 8/23/07, *Ed Pastore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi, folks.
I've been on the Asterisk Announce list for a while now, and it seems
to me that the release versions of Asterisk are a bit bleeding-edge.
They
Olivier wrote:
Hello,
I want to safely delegate ACD edition to a system administrator who has
no knowledge of Linux nor Asterisk.
More precisely, I want him to be able to edit and change menus such as :
Type 1 for management; 2 for support; 3 for sales department.
I could teach this
Doug wrote:
At 19:35 8/17/2007, Lee Jenkins wrote:
Bill Andersen wrote:
I'm a network admin that maintains 3 commercial Asterisk
servers for my employer.
I am wanting to move away from the pre-packaged commercial PBXs
to a more pure asterisk setup. The systems I have utilize
Bill Andersen wrote:
I'm a network admin that maintains 3 commercial Asterisk
servers for my employer.
I am wanting to move away from the pre-packaged commercial PBXs
to a more pure asterisk setup. The systems I have utilize a nice
web GUI to make changes, but it really limits what I can
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error
on channel 'Zap/3-1'
When this
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 15.08.2007, 10:14 -0400 schrieb Lee Jenkins:
Hi everyone,
I have been dealing with a certain issue with a particular customer site
for months now. The problem occurs when there is an error with caller
id as shown in the following:
WARNING
Matthew Harrell wrote:
Hi. I've got a working dial plan on my home system but there are problems
with it and I was hoping someone more comfortable with dial plans might be
able to help. In a nutshell here's what I'm currently doing on an incoming
outside phone call
[default]
satish patel wrote:
dear all
is there any GUI application with support asterisk 1.2
version i am useing 1.2 and i have fine more about GUI base
configuration but i didnt got any GUI package for asterisk 1.2
If you're a windows user, you can also check out DialplanPro:
Julian Lyndon-Smith wrote:
Sorry if this appears twice - I originally sent it nearly 18 hours ago
and never saw it ..
I have a need to have a unique integer number that can be used by a
dynamic meetme room (I am wanting to redirect a call into a meeting
room, and need a unique number to
Watkins, Bradley wrote:
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in
OCOSA ListAcct wrote:
Does anyone know how to have an ad or announcement playing but in the
background play a MP3 file?
I think this would be done with the s extension and background
application but not sure how? Any help would be appreciated!!
We just used Audacity and blended
Chris Mason (Lists) wrote:
Lee Jenkins wrote:
I'd say that Micro is the MS of Restaurant POS. We replace their
systems regularly ;)
I'm curious what with?
www.datatrakpos.com
Notice that I didn't say en masse but yes, we do replace a few Micros
systems a year. Same thing with some
Tomislav Parcina wrote:
There is hotel application weary popular in Croatia - Micros-Fidelio.
Now I need to connect Asterisk with this application for purpose of
billing. Thing is that hotel would like to give customer one bill for
every service that he used while he was in hotel.
Has
equis software wrote:
Hi!
I want to have this behabior in my queue.
When a call come in, if there are unavailable agents or and busy agents,
the queue reject the call.
Thanks !
I think you need checkout:
Introduced right after the v1.0 release
If you wish to remove callers from the
equis software wrote:
Hi!
I want to have this behabior in my queue.
When a call come in, if there are unavailable agents or and busy agents,
the queue reject the call.
Thanks !
Also check out
joinempty=strict
...it's in the same article:
Tomislav Parcina wrote:
There is hotel application weary popular in Croatia - Micros-Fidelio.
Now I need to connect Asterisk with this application for purpose of
billing. Thing is that hotel would like to give customer one bill for
every service that he used while he was in hotel.
Has
Lee Jenkins wrote:
Hi all,
I'm having an odd problem with my polycom 301. I am initiating a call
to it with AMI Originate() function:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: dropped_conf=111
The to_meetme context
Arun Kumar wrote:
Hi
I already tried asterisk manager but Im not able to get status for each
queue member.
thanks
That must be a problem with your configuration. I get QueueMemberStatus
on my AMI interface (1.2):
Event: QueueMemberStatus
Privilege: agent,all
Queue: support
Alex Roston wrote:
Hi Everyone,
I'm planning my first asterisk box, and I'd like to know what SIP
providers everyone likes. Voipjet? Gizmo? Somebody else?
Thanks,
Alex
I've been using www.axVoice.com for about 9 months now with great
results. Quality is good, but communication
Lee Jenkins wrote:
I've been poking for the definition of QueueMemberStatus and all the
source file indicates is that it is a integer member of the member
structure.
Anyone know where I can find the CONSTANTS definitions?
OK, I didn't know this, but QueueMemberStatus returns the same
I've been poking for the definition of QueueMemberStatus and all the
source file indicates is that it is a integer member of the member
structure.
Anyone know where I can find the CONSTANTS definitions?
--
Warm Regards,
Lee
___
--Bandwidth
satish patel wrote:
dear all
I am new in asterisk and i have now setup asterik for
40 phone now i want to configure call transfer between phone so how it
is possible and what configuration part in asterisk will perfomed for
this task give me suggestion for my solution
Zeeshan Zakaria wrote:
I had the same situation and I had to replace my T1 card with the one
with hardware echo canceller. All other solutions were failed. May be
you need to do the same if you're on a PRI or using PSTN lines. If
you're on a pure VoIP network, then its the phones.
On
satish patel wrote:
Dear ALL
I want to transfer call from one phone 2 another
phone so this is asterisk feature or SIP Phone feature or endpoint
feature how can i transfer phone call from to another phone
Rgd
Satish patel
Check out this page:
Jason Backshall wrote:
Do you have CallProgress=yes in your zapata.conf? This one just bit me
in the arse this morning. I set it to no and one-way audio went away.
Have heard of issues similar to this - and whilst disabling callprogress may
make that symptom disappear, it probably
Don Briggs wrote:
I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way
audio. A caller from the pstn world hits the tdm400 card, This rings two
phones in a ring group. My client answers the phone, the calling party is
told the customer here her but she can not here
Hi all,
I'm having an odd problem with my polycom 301. I am initiating a call
to it with AMI Originate() function:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: dropped_conf=111
The to_meetme context is very simple:
[to_meetme]
Dave Miller wrote:
Lee Jenkins wrote on 6/19/07 9:56 AM:
Vadim Berezniker wrote:
Enable verbose logging for the asterisk log
Set verbose level to 4
Review the log file for anything that looks like a phantom call.
There should be enough information to get some idea of why this is
happening
]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
Jenkins
Sent: Monday, June 18, 2007 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Phantom Calls
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls
]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee
Jenkins
Sent: Monday, June 18, 2007 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Phantom Calls
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls
David Boyd wrote:
Two points,
first (I believe from many previous threads, and viewing source code
) you must answer a call to place audio on the channel.
second, depending on the type of access being used by the originator of
the call, the carrier will not pass audio on the channel
Nitesh Divecha wrote:
Is there any info on how to create .call files with some examples? And
where to place this file? And how to initiate it..?
Thanks man...
Cheers,
Nitesh
Christopher Robinson wrote:
That should be pretty easy to do with a .call file. The context that
you
Hi all,
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only silence and then they hang back up
Stephen Bosch wrote:
Lee Jenkins wrote:
I have a client that is having problems with phantom calls. I have not
been able to see it happen myself, but they say when it happens, the
display on the phone (polycom 301's) says Device is calling, but when
they answer the phone, there is only
Matt wrote:
I too have seen what Rob is saying.. on a Sangoma card. It was an easy
fix in the config, but I don't remember what it was.. but basically it
was stray voltage.
On 6/18/07, * Rob Schall* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
We were having phantom calls
[EMAIL PROTECTED] wrote:
On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out):
Can't comment on this one, as I never use AGI to dial.
My AGIs just set the context, extension and priority,
and exit to the dialplan to do any dialling.
Lee Jenkins wrote:
Hi all,
My company has pretty much standardized on Polycom phones and I am in
the beginning phase of writing a GUI for administering/managing polycom
provisioning at multiple sites which we intend to release as OS. I've
started studying the docs and I'm having trouble
Doug Lytle wrote:
Steve Finkelstein wrote:
Hi all,
I have the following in my extensions.conf:
I use the mysql addon and create a subroutine that checks for black
listed numbers. I then call it at each entry point (For faxes as well):
; **
; Auto attendant
;
Kenneth Padgett wrote:
My company has pretty much standardized on Polycom phones and I am in
the beginning phase of writing a GUI for administering/managing polycom
provisioning at multiple sites which we intend to release as OS. I've
I'd love to be notified when you release the Polycom admin
programming. On the other side, I have no doubt that with an application
server and FastAGI you can get quite a lot of bang for the buck. :)
l.
On Fri, 08 Jun 2007 18:07:50 +0200, Lee Jenkins [EMAIL PROTECTED]
wrote:
We have found that generally speaking, running the FastAGI server
Kyle Sexton wrote:
I wrote an introduction to AGI programming paper as an exercise to learn
more about the process involved. You can find a copy of it here
http://mocker.org/papers/. I welcome any comments or corrections to
improve upon it. As I said, it was mainly done to force myself to
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