-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I've got Asterisk SVN-trunk-r9059 currently running on Fedora Core 4
w/ 2 eyebeam softphones and 2 Grandstream GXP-2000. At my desk I've
got the grandstream and the GXP-2000 I would like to ring both. Using
macro-exten-vm and dialparties.agi Macr
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
Too funny. But true.
The first time I did a 'show dialplan' after trying out AEL, I felt like
I was seeing an assembler dump of C++ :-)
-Johnathan
___
--Bandw
That is a good argument. But I am not sure yet. Do you know if there are big voice quality differences between the Digital and the Analog card? HousiRobert Webb <[EMAIL PROTECTED]> wrote: On Wed, 15 Feb 2006 08:59:22 -0800 (PST)housi mueller <[EMAIL PROTECTED]>wrote:> Hi there,> > I would lik
On Tue, Feb 14, 2006 at 05:14:36PM -0600, Bob McDowell wrote:
>
> Speaking of script launching, how would one fire-and-forget a script in
> Asterisk? It seems that as it is currently configured, if the called
> hangs up, the script aborts.
System(something &) ?
Or did you refer to AGI?
--
Tza
On Wed, Feb 15, 2006 at 01:41:59PM +0100, Koopmann, Jan-Peter wrote:
> On Wednesday, February 15, 2006 12:42 PM Garth van Sittert wrote:
>
> > Could we possibly see your settings to get this right? I am trying
> > to get it working at the moment.
> > I can see the phone buttons have subscribed
any one have this issue:
http://bugs.digium.com/view.php?id=6356
--
Peng Yong
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Whats wrong with the [EMAIL PROTECTED] wake up configurations?
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Dovid Bender
> Sent: Wednesday, 15 February 2006 4:54 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> S
Agreed. Like I said, I need to learn the rules for
Makefiles and provide a complete
solution. I'll make time next week to try to improve
the situation.
My logs show hundreds of downloads, yet only a very small
number of people have
emailed me questions, or provided suggestions for
improv
On Wed, Feb 15, 2006 at 09:29:39PM +, roswel ajf wrote:
> hi,
>
> To post, and to reply to a post, i have to goto my email. Now, if there is
> a web interface to these mailing list, things would be easier.
Your email client?
I don't know how Hotmail is. Some people even do the mistake oof u
Can the new echo
cancellation features in Speex 1.1.9 and higher be activated when using the
codec within Asterisk?
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The Makefile patch is only a couple of lines, so I
recommend patching it by hand.
I need to look into setting up an out-of-tree makefile that
is self contained, like
the modules in asterisk-addons, but I haven't had time to
tackle that yet.
I suspect that you managed to get it compiled, b
James Texter wrote:
I'm hooked up to a regular analog POTS line. I've tried both loop
start and ground start, but no luck either way. Any other thoughts?
Unfortunately, no. I only have experience with the Adit 600
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty t
Thanks :)2006/2/14, Eric ManxPower Wieling <[EMAIL PROTECTED]>:
Reli Loin wrote:>> hello,>> I have one account i need using multiple sipura ata, for my account.>> it's possible in asterisk.No. Generally you never need multiple devices to use the same account
information. This has been talked abou
Most/all assemblers have a better and more consistent parser though.
The parser for the extensions dialplan is just short of insane
John Novack
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
-Original Message-
From: Anthony Rodgers [mailto:[EM
I'm hooked up to a regular analog POTS line. I've tried both loop start
and ground start, but no luck either way. Any other thoughts?
Thanks,
James
Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
So, I'm still having this problem with outbound calls not working
when using a channel bank. I'v
Paul Hales wrote:
From memory, it's really down to making the right selections in sip.conf
We did a large installation, with phones at the Head Office using g711
and phones at remote sites using g729.
Asterisk happily transcoded for us. Which was great.
It keeps telling me (for g711 to G729
Darren Wright wrote:
You may want to turn the Rx gain down a bit..
-Darren
I'll give that a try, thanks.
Doug
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety."
_
[EMAIL PROTECTED] wrote:
So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 po
Douglas Garstang wrote:
Yes, programming the dialplan is akin to programming assembler.
Sorry, have to do it...
Hahahahahahehehehehaha. That should be a tagline.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve nei
Assembly instruction sets usually have more documentation.
Anyway, the OP has a hotmail address so he already has a "web interface"
to use for the list.
Douglas Garstang wrote:
>Yes, programming the dialplan is akin to programming assembler.
>
>-Original Message-
>From: Anthony Rodgers [
Yes, programming the dialplan is akin to programming assembler.
-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 15, 2006 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] is there a web interface t
I have [EMAIL PROTECTED] 2.5 running in my network. VEGA 50 Analog is used
as the gateway between PSTN & VoIP. I am able to dial out to PSTN with out
any issues. When a PSTN caller calls to my VoIP network he is able to reach the
extensions. But even after the caller (PSTN) disconnects, the
Hi
you can download the file here
http://kvin.lv/pub/Cisco/ata_03_01_00_sip_040211_1.zip
bye
On 2/15/06, Weiming Jiang <[EMAIL PROTECTED]> wrote:
Hi, I want have upgrade my ATA186 v2.15ms to H.323/SIP ButI dont have a cisco acount yet can some body help me with the ata18x-v2
You'll likely find Asterisk itself even more of a challenge then.
On Feb 15, 2006, at 1:29 PM, roswel ajf wrote:
hi,
To post, and to reply to a post, i have to goto my email. Now, if
there is a
web interface to these mailing list, things would be easier.
___
Doug,
The TFOT book recommends using expressions instead of dialplan
functions:
exten => s,1,Set(mainLoop=$[${mainLoop} + 1])
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, February 15, 2006 2:10 PM
To: Asterisk Us
I would've expected a channel bank or two to be the definitive solution.
With a T-1 connection or two back to the * server.
Michael Graves
Sr Product Specialist
Pixel Power Inc
[EMAIL PROTECTED]
o(713) 861-4005
o(800) 905-6412
f(713) 864-8668
c(713) 201-1262
> Original Message
Does this release possibly support japanese caller id?
Cheers
The Asterisk Development Team wrote:
The Asterisk/Zaptel development team is pleased to announce the release
of Zaptel 1.2.4.
This release contains a number of bug fixes, along some with new
functionality:
* The driver for the Xor
The Asterisk/Zaptel development team is pleased to announce the release
of Zaptel 1.2.4.
This release contains a number of bug fixes, along some with new
functionality:
* The driver for the Xorcom Astribank has been incorporated into this
distribution. Xorcom will provide primary support and driv
On Wed, 15 Feb 2006, Olivier Krief wrote:
Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3.
http://www.snom.com/firmware.html#1641
5.3.3 is not available for public download...
-Dan
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Anybody out there using GR303? Latest grumblings on the list are from
last spring. I'd like to use Asterisk as a concentrator/DLC speaking
303 to a 5E. Threads from awhile back mentioned support only for
Asterisk spekaing to a concentrator, rather than acting as one itself,
but the gr303fx
Freddi,
I started out this morning try to proof the concept of having Asterisk call an
AGI script, set several variables, and then return control to the dialplan
where it would execute the command. I wanted to set a number of variables in
the AGI for each number to dial. The first variable woul
The Asterisk/Zaptel development team is pleased to announce the release
of Zaptel 1.2.4.
This release contains a number of bug fixes, along some with new
functionality:
* The driver for the Xorcom Astribank has been incorporated into this
distribution. Xorcom will provide primary support and driv
What's the best way to increment a numeric variable in the dial plan?
I tried this...
exten => s,1,Set(mainLoop=${MATH(${mainLoop}+1)})
but that converts it to a floating point number (WHY???), so I end up with
1., which later on means I have to perform string manipulation to get rid
of the
Just realized I should have replied in private and not
to the list. Sorry in advance.
Dovid
--- Dovid Bender <[EMAIL PROTECTED]> wrote:
> I have a half system thats almost done. Had a client
> that wanted it then backed out. Please contact me
> off
> list.
>
> Dovid
>
> --- Michael Sampson <[E
when you say support asterisk do you mean IAX only or
sip as well ? I am in a little rush here but I can
write you a pretty big list of SIP providers that are
known to be good.
--- andrew matthews <[EMAIL PROTECTED]> wrote:
> http://connect.voicepulse.net
>
> They support astrisk, with iax2 :)
>
I may be missing something here but why wouldnt ATA's
work ? (other than cost).
--- maka <[EMAIL PROTECTED]> wrote:
> hello,
>
> I am planning a fairly large hotel VoIP system,
> using analog phones. It will
> consist of about 100 analog phones, that must have
> access to a VoIP server.
> I am co
Brent Torrenga wrote:
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the same
Brent,
The last time I was having random disconnects, it turned out to be that
I had busydetect=yes on my zapata.conf. I chan
Perhaps you can scale him back at the 79XX? Not only might it solve the
problem, but I'll bet the people talking to him on the other end would
appreciate it as well...
On the Aastra 9133i, for example, you can provide gain settings in the
(mac).cfg file.
Bob McDowell
-Original Message
I have a half system thats almost done. Had a client
that wanted it then backed out. Please contact me off
list.
Dovid
--- Michael Sampson <[EMAIL PROTECTED]> wrote:
> Does anyone have any system in place that does
> automated wake up calls.
> With recordings and options configurable over the
>
I am using an Asterisk box as a mini-softswitch and have run into a
minor (hopefully) road block. The far end switch requires CIC (Carrier
Identification Code) in the SIP invite like:
INVITE sip:+18001234567;[EMAIL PROTECTED];user=phone SIP/2.0
^^^
I
http://groups.google.ca/group/Asterisk-users?hl=en
-Original Message-
From: roswel ajf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 15, 2006 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] is there a web interface to this mailing list?
hi,
To post, and to re
On Wed, February 15, 2006 22:35, Brent Torrenga said:
> I have one use on our PBX who has been experiencing seemingly random
> disconnects. The user is on the same LAN as everyone else, using the same
> type of phone (79XX loaded with SIP firmware) as everyone else. He had
> some
> disconnects a fe
So, I'm still having this problem with outbound calls not working when using a
channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to
make sure it wasn't an equipment problem. I am using a Digium TE411P card, and
have simplified it down to just 1 port plugged into the cha
I have one use on our PBX who has been experiencing seemingly random
disconnects. The user is on the same LAN as everyone else, using the same
type of phone (79XX loaded with SIP firmware) as everyone else. He had some
disconnects a few weeks ago, I suspected the phone, so I swapped his with
mine.
hi,
To post, and to reply to a post, i have to goto my email. Now, if there is a
web interface to these mailing list, things would be easier.
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Jayson Navitsky wrote:
See the problem is when I do
Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL
PROTECTED]&Local/[EMAIL PROTECTED],30)
If someone is on the phone it returns Busy and then kills the incoming
call. ChanIsAvail would work great if I was going out to the PSTN
On Tue, Feb 14, 2006 at 10:17:16AM -0500, Jayson Navitsky wrote:
> Hi,
>
> So I've done my research on Chanisavail, read the wiki, checked the
> archive but can't seem to find anything to suit my scenario. I've
> played around with it a lot, but I'm still scratching my head on what
> I need to do
Hi All!
I need some feedback about the edge-core sip phones, somebody uses it?
They are reliable?
What the community say about them?
Miklos
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Hi Guys,
This article was posted few days back. I thought i can get more info here.
I am trying to bridge two outbound calls together. (have a program start a
context, dial one party and then bridge another party)
I thought that the G() flag in the dial application would work.
I tried the the
Well, netfilter is a decent firewall :). Give the sip-conntrack helper a try,
and then please tell me what u found.
see: www.iptel.org/sipalg for help.
Cheers.
Mensaje citado por: \"Koopmann, Jan-Peter\" <[EMAIL PROTECTED]>:
> On Wednesday, February 15, 2006 1:59 PM John Jensen wrote:
>
> >
Well, netfilter is a decent firewall :). Give the sip-conntrack helper a try,
and then please tell me what u found.
see: www.iptel.org/sipalg for help.
Cheers.
Mensaje citado por: \\\"Koopmann, Jan-Peter\\\" <[EMAIL PROTECTED]>:
> On Wednesday, February 15, 2006 1:59 PM John Jensen wrote:
>
On Wed, Feb 15, 2006 at 01:28:39PM -0500, Wojciech Tryc wrote:
> Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today
> announced that they have integrated PIKA’s high-density analog computer
> plug-in boards with the open source Asterisk PBX, with the introduction
> of PIKA Connect fo
The silence suppression is a client setting. Asterisk does not have
silence suppression as far as I know.
Garth
Dan Elder wrote:
Hi all, I'm getting some noise gate like effects on our sip lines & I think I need
to disable silence supression, I'm searching docs & not finding where this can b
In the latest CAVP conference call, the membership body voted to restrict
membership to VoIP LEC's and to create a seperate membership body for any
other parties interested in contributing to the CAVP's efforts in CRTC
lobbying and providing a unified industry presence in the Canadian telco
industr
Quoting andrutto <[EMAIL PROTECTED]>:
> I just want to ask if anyone has some experience with Alarmreceiver
> application in * 1.2? Is this
> application reliable (according to wiki it isn't)?
I don't see anywhere in the wiki where it says this is unreliable. The wiki
mentions that This
applic
Wojciech Tryc wrote:
Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today
announced
Take this to the -biz list... This is for asterisk discussion, not
marketing.
Jeremy McNamara
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Hi,
I just want to ask if anyone has some experience with Alarmreceiver application
in * 1.2? Is this application reliable (according to wiki it isn't)?
I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it
behaves very strange. Sometimes alarmreceiver is able to get so
Hello,
The astGUIclient web-client does most of this, it is open source and
entirely web-based so no need for JAVA:
http://astguiclient.sourceforge.net/
MATT---
On 2/14/06, Arne Morten Johansen <[EMAIL PROTECTED]> wrote:
> Hi there. We're going to develop a call centre app for internal use in
>
The
patch you saw is not for the stable branch.
Salu2
Jsalas
-Mensaje original-De: Moises Silva
[mailto:[EMAIL PROTECTED]Enviado el: Wednesday, February 15,
2006 2:28 PMPara: Asterisk Users Mailing List - Non-Commercial
DiscussionAsunto: Re: [Asterisk-Users] asterisk sil
Hi Dan,
How is your echo can the issue?
Did you disable the echo can and solve the DTMF issue? If you did,
did it trade the DTMF issue with echo problem?
It would nice if you can share your experience.
Thanks.
Andy
On 2/14/06, Dan Elder <[EMAIL PROTECTED]> wrote:
> Please ignore my last quer
Does anyone have any system in place that does automated wake up calls.
With recordings and options configurable over the phone?
--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000
___
--Bandwidth an
Ottawa, Canada – February 15, 2006 - PIKA Technologies Inc. today
announced that they have integrated PIKA’s high-density analog computer
plug-in boards with the open source Asterisk PBX, with the introduction
of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software
layer, available f
Asterisk DOES NOT HAVE silence suppression (VAD) support for now. So it
cannot be disabled or enabled. Simply does not exists. A couple of
weeks ago i saw a patch to enable it. The link here:
http://bugs.digium.com/view.php?id=5374
so unless you have the previous patch, you should disable silence
Paul A. Pringle wrote:
> Occassionally on calls we get what sounds like low volume channel
> bleedover. Not clear enough to make out words, but not echo of either
> side of the main coversation. We're using a Digium card with 11
> channels connected to PSTN lines. Any ideas on what the problem i
I dont recall the
SPA-941 playing a stutter tone in the previous firmware but it is driving me
nuts, anyone know where to turn it off?
Kerry GarrisonDirector of
Technical ServicesTech Data Pros - Orange County's Mobile IT Service
Provider(949) 502-7819 x200 - [EMAIL PROTECTED]http://www.tec
On Wednesday 15 February 2006 12:49, Paul A. Pringle wrote:
> Occassionally on calls we get what sounds like low volume channel
> bleedover. Not clear enough to make out words, but not echo of either
> side of the main coversation. We're using a Digium card with 11
> channels connected to PSTN li
Hi All
Has anyone managed to get the hint priority with Swissvoice IP10S phones
working?
I have 2 phones: a Snom 360, setup as the reception phone on extension
11, and a Swissvoice IP10S on extension 12.
When calling each other (tested both ways) I can only ever see the Snom
360 in the Active
I've had pretty good luck getting the telco to bring out a laptop and
test the lines for this sort of thing. Not past the DMARC, of course,
but still it helps to narrow problems down.
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul A.
Occassionally on calls we get what sounds like low volume channel
bleedover. Not clear enough to make out words, but not echo of either
side of the main coversation. We're using a Digium card with 11
channels connected to PSTN lines. Any ideas on what the problem is or
how to go about troublesho
Hi
This may be off topic because it involve cable.
I am testing with Arris cable modem / MTA
I have 2 models, one older and one newer.
With older one, everything works fine
With the new one, I can register, make a call and I hear the other person
but he can't hear me
The config is the same
Hunt, Bill wrote:
> I would recommend that you look at the Pika Technologies Daytona MM
> board. It has onboard DSP and onboard analog bridging taking up much
> less horsepower. Please contact me off-list if you would like more
> information.
>
> Bill Hunt
> Stroudwater Contact Point
This list is
On Wed, 15 Feb 2006 08:59:22 -0800 (PST)
housi mueller <[EMAIL PROTECTED]> wrote:
Hi there,
I would like to connect an Aasterisk Server with a
Panasonic PBX (has E1extension).
I only need 4 Lines. So I thought I could use an
Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1
card wh
Nik,
Looks like you're making some progress. When I first started using [EMAIL
PROTECTED]
I had trouble getting the outbound dialing to work. I wasn't sure where
to start, so what I did was skip the macros in the dial plan. I wanted
to play around with exactly what digits the telco wanted to s
Have some NMS TX4000-4link Full stack for sale.
Mark
www.voiceinternational.com
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Hi there, I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive. I dont now which card to take. Please tell me what you thi
Garth,
Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3.
Reagrds
- Original Message -
From: "Garth van Sittert" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 15, 2006 12:41 PM
Subject: Re: [Asterisk-Users]
Hi,
Anybody from Québec wanting to get there with me ? I have 2 places left
in my car for those who want to share the ride.
Thanks,
Michel Belleau
SERVICES INFORMATIQUES MALAIWAH.COM
(418) 261-6412 -- http://www.malaiwah.com
Adrien Laurent a écrit :
>Hi,
>
>This is a reminder about our next
Hi,
This is a reminder about our next meeting.
It will be held from 6pm to 8pm, February 21 at Modulis offices which
are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal.
Thanks to Claude Patry, we will be having a 20 minute conference call
with Mark Spencer.
If you'd like to ask Mark a
See the problem is when I do
Dial(Local/[EMAIL PROTECTED]&Local/[EMAIL PROTECTED]&Local/[EMAIL
PROTECTED]&Local/[EMAIL PROTECTED],30)
If someone is on the phone it returns Busy and then kills the incoming
call. ChanIsAvail would work great if I was going out to the PSTN
looking for a channel, bu
Hi all, I'm getting some noise gate like effects on our sip lines & I think I
need to disable silence supression, I'm searching docs & not finding where this
can be set, does * have a setting to turn this off? basically what's happening
is when we stop talking, the other end hears total silence,
You may want to turn the Rx gain down a bit..
-Darren
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Joseph Tanner
> Sent: Wednesday, February 15, 2006 10:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Su
Joseph Tanner wrote:
Shouldn't hurt, I'd give it a try. But first you may want to fiddle
with the Tellabs configuration some more. This has some good
information:
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
I know, I've lived on that page during the setup of the ca
quadrasoftware.com has the same app. its open source.On 2/15/06, Lenz <[EMAIL PROTECTED]
> wrote:Hi Arne,what you write about seems to be mostly what Flash Operator Panel does.
Check it out before writing a clone yourself! :-)l.On Tue, 14 Feb 2006 13:21:37 +0100, Arne Morten Johansen <[EMAIL PROTEC
Currently, with default settings only outgoing calls are recorded. How can
I enable inbound?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: 15 February 2006 15:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Aste
> My 5 cents worth is if you use Bristuff stable you must use
> Asterisk-1.0.10 ( Old version ) If you use Bristuff 3PRE1l
> you will have problems with FXO cards as I did.
> Bristuff3PRE1l is not Stable use at own risk!!!
Can't speak for anyone else, but we have 2 sites running HFC cards with
B
Shouldn't hurt, I'd give it a try. But first you may want to fiddle
with the Tellabs configuration some more. This has some good
information:
http://www.voip-info.org/wiki/view/Tellabs+Hardware+Echo+Cancellers
Joseph Tanner
On 2/15/06, Doug Lytle <[EMAIL PROTECTED]> wrote:
> Since putting my
Since putting my Tellabs EC into place around 2 weeks ago, the echo
problem has almost been eliminated. Reports of some very faint echo,
but everybody is happy.
My question is, if I were to also turn on the Asterisk Software EC,
would this remove any residual echo that may make it past the Te
Andres,
Thanks for the explanation!
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Andres
Enviado el: miércoles, 15 de febrero de 2006 1:31
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Telmex PRI line configu
Hi,
I am running a call center based on Asterisk and
building some statistics based on the queue_log file.
I have some doubts about it that maybe you could
help (actually, maybe these doubts are suggestions for
enhancements!):
1st Scenario - Agent receives the call, and puts it
on parkin
Hello,
Currenly I've [EMAIL PROTECTED] 1.5 running on DMZ. I can register SJphone
there, good audio on 8200 (webmeet me calls) and i also can dial
Zapata extensions.
When I dial sip phone extensions nothing happens if the client that
i'm calling is registred, if the client has voicemail it goes
Hi Arne,
what you write about seems to be mostly what Flash Operator Panel does.
Check it out before writing a clone yourself! :-)
l.
On Tue, 14 Feb 2006 13:21:37 +0100, Arne Morten Johansen <[EMAIL PROTECTED]>
wrote:
Hi there. We're going to develop a call centre app for internal use i
-- Forwarded message --
From: Marco Mouta <[EMAIL PROTECTED]>
Date: Feb 15, 2006 1:58 PM
Subject: Which ATA device do you recommend?
To: [EMAIL PROTECTED]
Hello,
I'm developing a Voip Solution for a client, which ATA SIP do you
recommend? there are some ATA devices fully tested w
James Steven wrote:
Hi
What is the easiest method to set up CDRs for inbound calls? Can this
be achieved without use of AGI and programming?
Thanks for your help.
James
if I am not misunderstanding you, CDR's are automaticall written for
ALL calls through the system. to specefically hand
>Could we possibly see your settings to get this right? I am trying to
>get it working at the moment.
>I can see the phone buttons have subscribed to asterisk, but they just
>don't light up. We are using 4.1 firmware and are upgrading to 5.3 to
>see if it helps.
Working good here in the Great
On Wednesday, February 15, 2006 1:59 PM John Jensen wrote:
> Hi Hagen,
> It's not exactly a pleasure to run SIP through firewalls but it can
> be done.
> At least in under some circumstances.
If you use a decent Firewall it will analyze and interpret the SIP Headers
etc. and open the correct po
Hi
What is the easiest
method to set up CDRs for inbound calls? Can this be achieved without use
of AGI and programming?
Thanks for your
help.
James
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It works!I hadn't put the rule for app_cbmysql.so: app_cbmysql.o.Not really easy to install on * 1.2.4 for non-dev people (as the patch makefile doesn't work). Thanks you very much Sean and Dan.
On 2/15/06, Sean Cook <[EMAIL PROTECTED]> wrote:
-BEGIN PGP SIGNED MESSAGE-Hash: SHA1You have to
hi all,
hope any one can help create a trunk, i'm talking to a voip gateway provider
right now, they gave me the IP address of their server a prefix to
authenticate calls. How can i create such a trunk? example prefix is 1234#
and IP address is 1.1.1.1, in ser i was able to do it by just simply
re
I am assuming you made a profile in sip.conf like so
[sipdevice]
type=peer
host=x.x.x.x
...
.
.
disallow=all
allow=ulaw
and in extensions.conf
exten => _X.,1,Dial(SIP/sipdevice/${EXTEN})
then this MUST work. :)
you can do a sip debug or set debug 10
yusuf
Matt wrote:
Hi,
How do I specify
98] logger.c: -- Executing
Macro("IAX2/206-4", "record-enable|206|OUT") in new stack
Feb 15 09:07:00 DEBUG[30698] pbx.c: Function result is '0'
Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Executing
GotoIf("IAX2/206-4", "0 > 0?2:4") in ne
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