We couldn't set our 1.0.10 Asterisk system to pickup calls with Snom phones.
I've read patches http://bugs.digium.com/view.php?id=5014 and
http://bugs.digium.com/view.php?id=5853 could provide that with 1.2.X but we
never tried ourselves.
I would be very happy to know if someone put that in a
Igor Neves [EMAIL PROTECTED] writes:
Hi,
Does anyone have any experience connecting asterisk to alcatel 4200
series pbx with bri cards?
Does it should work with asterisk bri in NT mode, and alcatel bri with
TE mode?
Hi Igor,
we are doing that. Bristuffed Asterisk with two HFC-cards is
I do not even know which brands/models to consider that are
out there. Given that we are in the US, and want to use BRI
to improve sound quality (no echo, no static), what would be
some good cards to look at? I hear a lot about BRIStuff,
which I think is used on the Junghanns cards (like
On 2/14/06, Michael Collins [EMAIL PROTECTED] wrote:
Nik,
I'm not sure that NOP is correct, but I'm in the states so I'll to
defer to someone who knows E1/PRI. When I run zttool I have OK under
the alarms. Is there a way you can call the telco and confirm the
settings? Make sure that
On Wed, 2006-02-15 at 08:47 +, Chris Bagnall wrote:
I do not even know which brands/models to consider that are
out there. Given that we are in the US, and want to use BRI
to improve sound quality (no echo, no static), what would be
some good cards to look at? I hear a lot about
with the following configuration:
zapata.conf
[channels]
language=it
context=from-pstn
signalling=pri_cpe
switchtype=5ess
rxwink=300
callerid=asreceived
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
Hi All,
I've just put together a system comprising of the following;
Hardware
2 x AMD Opteron 270 Processors (Dual Core)
Tyan K8WE Mobo
2GB Kingston PC3200 Registered RAM
2 x WD Raptor 1rpm 74Gb
Digium TE210p
Software
Mandriva 2006 Public Release (Kernel 2.6.12-12mdksmp)
On Mon, 2006-02-13 at 21:20 -0800, Michael Collins wrote:
JCC,
So let's consider an operator, takes a call and decides to attended
transfer it to Bob because it's slow and she want's to ask something,
but the instant she picks that option another call comes in. If
hanging up converted it
Hi,
I'm experiencing brief pauses during my calls: 0.5-1.0 sec of silence if
call
continues for more than a few minutes.
I'm sure that problem is in the phone (a cheap ATCOM AT-320 with latest SIP
firmware) but I'd like to diagnose better.
During a little test, it seems that there is no problem
On 2/7/06, Nabeel Jafferali [EMAIL PROTECTED] wrote:
Removing this line will likely fix the problem. Since you don't have a NAT,the qualify= setting doesn't help keep the port(s) open. At the same time,most SIP devices have a NAT Keep Alive option, if that is an issue.
HelloIt did fix my problem,
Subject: RE: SIP Register
From: Tomislav Parcina [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
First impressions telling me you want to check your phone settings. What
phone are you using and what are the config settings?
Hi Mark, thank you for your reply.
I'm
Subject: RE: Queue - check agent
From: Tomislav Parcina [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Hello,
I might be wrong here, but I thought that in Queues.conf, if you defined a
queue with joinempty=no, or joinempty=strict then no calls will be placed in
Does this work with asterisk 1.2.4?I can't make app_cbmysql work.I get an error when starting asterisk:[app_cbmysql.so]Feb 15 10:26:53 WARNING[7616]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_cbmysql.so: undefined symbol: mysql_store_result
Feb 15 10:26:53 WARNING[7616]:
Maybe to a voicemail message box, which then gets emailed to a special email
account.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tomislav Parcina
Sent: 15 February 2006 10:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Why do you think it's phone problem and not Asterisk? Asterisk is the
one that contents my provider. * is the one who should decide what
information's to send to my VoIP provider... Anyway, I'm inexperienced
with this and I'm just trying to understand what is happening and where
could be
We are using a PRI connection between Asterisk and an Alcatel PBX 4400.
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Igor Neves
Sent: Monday, February 13, 2006 11:13 AM
To: Asterisk Developers Mailing List; Asterisk Users Mailing
On Tue, Feb 14, 2006 at 03:09:59PM -0700, Chuck Bunn wrote:
Hi Giorgio:
That seems like a kind of a kludge. I would rather have the program work
right, than adding a work around. Dan of Littlejohnsconsulting has told
me of one problem in ARI that he is fixing but I do not understand how
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
The error looks like a problem with the MySQL libraries on
your system. I have not
tested it against 1.2.4, but do have it running on SVN 7668
and have had it running
on 1.2.0
I can try 1.2.4 next week if you are not able to resolve it
by them.
Dan
From: [EMAIL PROTECTED]
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.
Regards
Garth
Darrell Long
Hi
We are currently using Asterisk 1.2.4 with IAX and app_meetme for
conferencing, but are looking to move to SIP because of issues with an IAX
control we're using.
The reason we moved from SIP to IAX in the first place was because of the
poor NAT traversal with SIP. At that stage we were
I moved the card to a different pci
slot and that removed the error.
thank you!
Phil.
yusuf [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
14/02/2006 15:32
Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To
Asterisk
IMHO, the Diva Server BRI range of cards are worth considering. The Diva Server
4BRI card is an active card that can do echo cancellation, automatic gain
control etc. The 4 port card costs similar to 2 single port cards so there will
also be room to expand if you need it.
More information can
Hi,
I have two sites and I'd like to connect them with a IAX trunk and share the
dialplan. Extensions cannot be clearly separated. Do I need to use 'switch'
statement or DUNDI/e.164?
Using 'switch', does any user can call any extension on both sites?
Thanks
Mimmus
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!!!
Regards Allan Gee
Phone: +27 21 4644400 Ext. 103
www.equation.co.za
Hi
Thanks to all who had given advice , I had done
connection between 2 IAX server , I am able to dial
and communicate now , some of the problems which I
faced is that
when I tried to dial , it was searching was of
default one.
and I was getting message like
Rejected connect attempt from
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 considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card.
Has
Hi Mr Gee
I am using the Duxbury HFC PCI Bri card and found it to be very stable running
asterisk-1.2.4 with Zaptel-1.2.3 with bristuff-0.3.0-PRE-1 on FedCore 4
Only problem is that you can only have FXO OR FXS on a card and not both on the
same 1 port BRI card
Regards
Tertius Smit
Hi,thanks for your quick answer.My system is Gentoo with mysql 4.1.14 installed from oficial gentoo repository. And mysql does work for other applications (I also already created the meetme db/table).
Maybe the problem comes from my manual patching of the makefile to compile app_cbmysql.c (as the
Thanks for the reply. Neat ideas there, but a couple of issues.
1. Don't want to have to jump around between the FastAGI and the dial plan. Our
plan is to have NO customer data in the dialplan, as all data will be contained
within MySQL. We don't want to have to make _any_ edits to the dial
You can try to run "make" in the linux source-folder. I had
the same problem Running FedCore 4 on a Dual Xeon Server and running make fixed
the error
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
TeesdaleSent: 15 February 2006 12:19To:
Can somebody guide me on how to get the ss7 channel up and running?
I have read some information on the ss7 but I need to know which card is
better and I wouldn't mind the configuration options too
goksie
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Hello,
If you are doing that many analog extensions you might want to
consider 4 channelbanks and a quad T1 card instead(or two 2-port cards
in two servers). Four TDM24XX cards will draw a whole lot of power and
would be much harder to replace than an exterior channelbank if
something goes wrong
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 to asterisk, but they
just don't light up. We are using 4.1 firmware and are
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
207 347 8080 x219
877 870 1234 Toll Free
Hi Hagen,
It's not exactly a pleasure to run SIP through firewalls but it can be
done.
At least in under some circumstances.
I have successfull run an Asterisk server from behind a NAT router
and run a SIP trunk to the SIP VoIP provider. The problems tend to
arise when multiple SIP devices wants
Kyle,
Right... we have hookups to Intrado at the moment and are doing it for
our ATA customers. I just was trying to think if a Softphone would be
compliant. Everything I've thought of seems to indicate it would be,
but wanted thoughts from other people.
On 2/14/06, Kyle Hagan [EMAIL PROTECTED]
Dont know. All i know is that i had ulaw enabled in * and i was getting
errors relating to iLBC.
The first thing to check is whether the IAXy even does iLBC
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To
im having a problem running zap on astbill.
when i dial any number through zap, astbill should minus balance if the call
gets through but it minus balance even I cancle the call.
any1 running astbill experienced the same ?
onthe otherhand, billing on sip/iax interface is working fine, even
On Tuesday 14 February 2006 19:11, fabrizio wrote:
hi all,
I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.:
-- Executing Dial(SIP/2003-bbae, zap/2/03460816149|30|t) in new stack
Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel
type registered for 'zap'
Feb
We have 3 existing switches interconnected via dpnss, we need to
integrate asterisk with these switches via a dpnss link.
Any suggestions?
also does anyone have a link to the differences between isdn30 and dpnss.
Thanks in advance
Bails
___
On Wed, Feb 15, 2006 at 01:29:10PM +, bails wrote:
We have 3 existing switches interconnected via dpnss, we need to
integrate asterisk with these switches via a dpnss link.
Any suggestions?
also does anyone have a link to the differences between isdn30 and dpnss.
Get a DPNSS to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
If memory serves me correctly this has to do with ABE only supporting
that number of watched extensions. You are correct that this is an
artificial limitation and I think someone from digium actually
commented that this should be improved in the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
You have to link it to the mysql libraries... add the following to the
apps/Makefile
APPS+=app_cbmysql.so
app_cbmysql.o: app_cbmysql.c
$(CC) -pipe -I/usr/include/mysql -L/usr/lib/mysql $(CFLAGS) -c
- -o app_cbmysql.o app_cbmysql.c
Hi,
How do I specify a codec to use for a SIP call?
IE.. If I'm doing Dial(SIP/blah) for some reason the call is
connecting using the codec at the bottom of my allow list rather then
top (G711u)... and I'd like to force it to G711u if possible.
___
Hi i'm developing a solution with ASterisk, but in fact i don't know
which ATA SIP device should buy.
Could you give me some advices?
Marco Mouta
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
FYI... I am running this on 1.2.4 and trunk
Sean Cook wrote:
You have to link it to the mysql libraries... add the following to
the apps/Makefile
APPS+=app_cbmysql.so
app_cbmysql.o: app_cbmysql.c $(CC) -pipe -I/usr/include/mysql
:-((
There's nobody with any idea here? :-((.
I need to force * to not try native bridging, at least when there are
different codecs used.
In current config * tried native bridge, it fails, but CDR has been
already generated and writed :-((.
Thanks a lot for your time (and
: -- Executing
AGI(IAX2/206-4,
recordingcheck|20060215-090700|1140012420.22) in new stack
Feb 15 09:07:00 VERBOSE[30698] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Feb 15 09:07:01 VERBOSE[30698] logger.c:
recordingcheck|20060215-090700|1140012420.22: Outbound recording
I am assuming you made a profile in sip.conf like so
[sipdevice]
type=peer
host=x.x.x.x
...
.
.
disallow=all
allow=ulaw
and in extensions.conf
exten = _X.,1,Dial(SIP/sipdevice/${EXTEN})
then this MUST work. :)
you can do a sip debug or set debug 10
yusuf
Matt wrote:
Hi,
How do I specify
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
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
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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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 ports
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
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
-- 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
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 in
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,
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 parking
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
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
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
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
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:
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
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
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
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,
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, but
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
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
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
asterisk-users@lists.digium.com
Sent: Wednesday, February 15, 2006 12:41 PM
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 think
Have some NMS TX4000-4link Full stack for sale.
Mark
www.voiceinternational.com
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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
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
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 not
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
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
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
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
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
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
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 is
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
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
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
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 query
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
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
our
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
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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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
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
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 be
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 for
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:
Hi
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:
Hi
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
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