Good day,
I have configured a queue like this which is actually
working. What is not working is the drop out of the
queue by pressing a single digit. I press digits
but Asterisk does not react in any way. However, what
is clearly detecting is a * to hang up the call
so I doubt it is something
[EMAIL PROTECTED] wrote:
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten =
I'm guessing it's the sudafed that caused me to wildly try this, but I'm
glad I did, because though it creates a new concern, it solved my
problem. Just for kicks I tried setting the canreinvite parameter to no
for the broadvoice peer, and that fixed everything.
My server is on a live ip,
Hello
I am working on an Asterisk Enterprise Manager Eclipse plug-in.
It will allow you to edit the Asterisk configuration via wizards and will
have an editor for configuration files with some sort of syntax highlighting
and auto-complete.
It will allow you to make these changes remotely and
hello
i want to rout my calls to h.323. i have registered my
asterisk with GnuGatekeeper. but it is not routing my
call to h.323 channel. he is saying Internal channel
initialization failed. Bad binary?
can any one check my settings what is problem here
thanks in advance
kamran
On Mon, 14 Mar 2005 14:42:00 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Mon, 2005-03-14 at 17:31 -0300, César Davi Ávila do Nascimento wrote:
Talk about skype is forbidden, but to be impolite is allowed...
Great list!
Did you ask about skype in asterisk? Did you ask an asterisk
I have one and only one reason to use OH323, and that is it
works with Cisco's CallManager. The standard H323 channel
has one-way audio issues when connected to CCM and fixing them
has been identified as a low priority/never going to happen
task.
Dan
-Original Message-
From: [EMAIL
* vdasilva ([EMAIL PROTECTED]) ha scritto:
Hello
I am working on an Asterisk Enterprise Manager Eclipse plug-in.
Do you plan to release this in a way available to the public or
commercially?
cheers
--
Programmers are machine which turn coffee into programs.
I've been too busy to write up any instructions, but the general
process is this-
1. Download the source for MeetMe2.c from
http://www.areski.net/asterisk-meetme/about.php?s=0 and put it in
asterisk/apps
2. Download the source for CBMysql from
http://www.mithotech.com/asterisk/
3. Extract the
A speaker list is more advanced. Probably bet implemented with a
manager
interface to meetme that passes the dtmf digits to the manager
interface
and allows the manager interface to manipulate the conference.
A managable speaker list is part of the out-of-tree meetme2 application
and
PHP web
hello list (2nd try as my first post seems to have gone astray in the endless
realms of tcp/ip),
i searched for nearly a week for a solution to this problem, as there is:
analog fax machine -» grandstream ata -» asterisk -» sip trunk from provider
-» provider gateway to pstn -» analog/isdn fax
Guys.. Why is it that when a call comes to a call queue and in term gets
assigned to an agent, if that agent tries to xfer the call using # or any
other feature, it doesn't do anything? I just hear the pleeps on the phone
but asterisk doesn't intervene with the Transfer prompt.
Am I missing
I need to do load balancing only for the following functionalities:
1) Registration of SIP clients to * servers.
2) Load balancing of the INVITEs from SIP clients to different * servers.
I'm not interested in supporting the features, which you have
mentioned below. I'm not aware how the
I read that, Asterisk strips off anything after the @..., so the IPis removed.
Using the ${SIPDOMAIN}, the SIP call can be made directly, but how do I forward the SIP call to a proxy (like SER), retaining the IP part?.
I am using XLite.
Do you Yahoo!?
Yahoo! Small Business - Try our new
On Tue, 2005-03-15 at 03:53 -0600, Anton Krall wrote:
Guys.. Why is it that when a call comes to a call queue and in term gets
assigned to an agent, if that agent tries to xfer the call using # or any
other feature, it doesn't do anything? I just hear the pleeps on the phone
but asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi
How do you queue the incoming call ?
Do you queue the call with the t option (allow the called user to
transfer the calling user) ?
Regards
Joo Amaro
Anton Krall wrote:
| Guys.. Why is it that when a call comes to a call queue and in term
On Mon, 2005-03-14 at 22:30 -0600, Kristian Kielhofner wrote:
cdr_addon_mysql doesn't compile at all, no matter what the OS.
Really? so what would this be:
cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/include/mysql-c
-o cdr_addon_mysql.o cdr_addon_mysql.c
cdr_addon_mysql.c: In
We had the same problems with transferring calls in queues.
Sometimes, after pressing the # Key twice !!, we hear Allison say
Transferring.
Which Phones do you use?
What shows up in the cli debug?
Are you using t and T options in the dial command?
Regards,
Guido Hecken
Guys.. Why is it that
Hi there
Just a quick question. I have been playing around with asterisk CVS-1.0.02
on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v
1.0.6 but am still a little uncertain which linux kernel is best to run on
?, can I use Fedora Core 3 (is it the preferred kernel) or
On Mon, Mar 14, 2005 at 05:03:27PM -0600, Steven Critchfield wrote:
On Mon, 2005-03-14 at 16:46 -0600, Jay Milk wrote:
Here's a perfect example of a mis-post that was blown out of proportion.
When I first saw Cesar's post, I responded off-list to the effect that
this would be the wrong
* Raoul Bönisch [EMAIL PROTECTED] [2005-03-14 19:26]:
* Stu Gotz [EMAIL PROTECTED] [2005-03-14 16:56]:
The H0,H1 timing may be tricky, but, If the modem is AT compliant, ATD! is
the flash command. The timing is based on S register 29.
Yes, that's another possibility. We're close to it. I
On Tue, 2005-03-15 at 10:47 +, Brett, Gary wrote:
Hi there
Just a quick question. I have been playing around with asterisk CVS-1.0.02
on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v
1.0.6 but am still a little uncertain which linux kernel is best to run on
?,
Did you try enabling sip debug on Asterisk and checking what it tells
you ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Pushor
Sent: Montag, 14. März 2005 21:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Brett, Gary wrote:
Hi there
Just a quick question. I have been playing around with asterisk CVS-1.0.02
on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v
1.0.6 but am still a little uncertain which linux kernel is best to run on
?, can I use Fedora Core 3 (is it the
I'm having the same problem over here, but with both, inbound/outbound
calls, I use a SER server auth my users, and when I need to use a VoIP
line that is not at my server, I use Asterisk to auth line outside my
server at my Foreign Voip server then when I get the line I can dial, but
none of
Hi.
I have a problem which I assume would be easy to fix, but I can't find
anything about it...
I wish to have people dialing my phone, and if it is busy, they are put
into a queue. And then I am dialed back when the previous call is
finished, and connected to the waiting caller.
Easy enough?
We tried FC2 and FC3. There were a bunch of changes in FC3 that I don't
really like all that much. I've always had great luck with FC2 so that
is what we ultimately went with.
Brett, Gary wrote:
Hi there
Just a quick question. I have been playing around with asterisk CVS-1.0.02
on fedora core 1
Hi,
What does this line of output mean?
Bridge stops because we're zombie or need a soft hangup:
I'm seeing this sometimes... I've looked in channel.c,
but the code is not much more revealing than the
debug line...
--
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 3
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten = 8500,1,VoicemailMain
exten =
I just did this for a customer. All I did was create a queue just for him,
he is the only agent in the queue. * acts just like you want if you are the
only person in the queue.
- Original Message -
From: Jan Marius Evang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
I've read several of the Wiki sections on Voicemail and Asterisk
variables but could not find an obvious answer to this question.
I would like to run a script that post-processes the voicemail after the
Voicemail application returns (with AGI or DeadAGI), but I cannot figure
out how to easily
Hi. Is this different depending on wether I am a Member or Agent?
I have used Memeber and it does not work...
Yours
Jan Marius Evang
I just did this for a customer. All I did was create a queue just for
him,
he is the only agent in the queue. * acts just like you want if you are
the
only
Hi,
Ive been searching the lists and cant find the exact solution I need using *. I need to route voice channels between two sites across cisco routers.Both PBX's are analog only (noDigital upgrade path).
I was thinking for the Gateways at each site.Have 2 FXS/FXO in each gateway.Two ports can
my $sth = $dbh-prepare(SELECT * FROM routes WHERE . $dbh-quote($num
ber) . RLIKE pattern ORDER BY LENGTH(pattern) DESC);
Does it mean I just need to use:
^61.* 100
^6178.* 150
^615.* 130
^61342.* 180
Ronald,
The ASTCC sql SELECT used will return the routes entry that matches the
Hi!
Is it possible to use Asterisk for web-triggered calls? For example, if
I have IAX termination from VoipJet and I want to tell Asterisk to call
two phones and when the calls are answered, connect them with each
other. I would like to offer this to end-users on a prepaid basis.
If it is
Great Steven !!!
Thank you very much. I installed the libtiff-devel package and all run
peacefully !!!
KInd regards,
Juanjo
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To
Quoting [EMAIL PROTECTED]:
Date: Mon, 14 Mar 2005 22:23:54 -0700 (MST)
From: Greg Hill [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] How NuFone.Net's customer service
works.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
Hi pros,
Newbie to asterisk, need some help.
My existing senerio is we have 6 analog quintums and 1 digital H323,
and our gatekeeper is gnugk openh323 located in US.
Our business is Call Center and our method of dial is using prefix and
gateway IP provided my Carrier.
I also brought two
I made the ISA/jumpers comment just the other day. God I miss the old days.
On Mon, 2005-03-14 at 19:26, Ron Joffe wrote:
On Monday 14 March 2005 16:18, Eric Wieling wrote:
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a
few SIP
Good Day list,
I am having
some issues with my card in that it wants to share IRQs
with everything else in my box.
I am
running WhiteBox Linux 3.0
Is there a
way to tell linux to assign a specific IRQ to a card. (unfortunately my MB does not
have feature of assigning IRQ to
Actually, not a perfect example:
In this case there was exactly one poster who answered on-list to the
original post. The answer was wrong list. The original poster insisted
then on re-asking the question, and then the flames really started.
Which is why I think the way of doing it is
I've put my fire proof suit on ready to ask a question ;o)
Redhat 9
Kernel 2.4.20-8
AVM Fritz PCI
Problem 1...
i have followed instructions from:-
http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
(Plus many many
I'm not a PRI expert and therefore don't know what this debug stuff means
for PRI, so if anyone can help me here...
I'm running the latest libpri and zaptel from CVS.
Keep in mind that everything works fine when using the STABLE libpri and
zaptel.
I am NOT running CVS asterisk. I am running 1.0.6.
Jed Stafford wrote:
Has anyone had any luck getting the addon's to compile
under OS X? I have been able to get 1.06 to build and
run great, but I really want to get the
cdr_addon_mysql.so to enable mysql writing.
When I try to compile I get a bunch of Undefined
symbols and I have tried
You still have to send another real phone number to the gateway
and tell the gateway what to do with it...
I just don't see it in your configs
Brgs
Tomek
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I just made this work without much difficulty, so I imagine you can do so
too.
It does not look like anything wrong with your queue entry.
Can you post your [testqueue-drop] context from extensions.conf?
/edg
--On Tuesday, March 15, 2005 9:02 AM +0100 Sascha E. Pollok
[EMAIL PROTECTED] wrote:
Juanjo Portela wrote:
Great Steven !!!
Thank you very much. I installed the libtiff-devel package and all run
peacefully !!!
KInd regards,
Juanjo
The installation instructions merely said you need libtiff installed. I
just changed them to say that if you use RPMs you need the libtiff and
Try adding an agent and logging into that agent (rather than making the SIP
phone the member, directly).
/edg
--On Tuesday, March 15, 2005 1:19 PM +0100 Jan Marius Evang
[EMAIL PROTECTED] wrote:
Hi.
I have a problem which I assume would be easy to fix, but I can't find
anything about it...
I
I used agentcallbacklogin app with a extension macro, that way if he was on
a conference call he could log out and the caller would be sent to his
voicemail. Here is another approach though, it is in the WIKI and does
work
http://www.voip-info.org/wiki-Asterisk+tips+campon
- Original
This question really has no one reply. The different Linux builds all have
there reasons. If your used to Fedora Core 1 then that is what you should
use. I use CentOS which is a clone of RHEL 3. They have just released there
Version 4 which is based on RHEL 4. It works and since I am used to
look into .call files, check the wiki for this:
www.voip-info.org
On Tue, 15 Mar 2005 15:16:14 +0100, Alexey Goloshubin [EMAIL PROTECTED] wrote:
Hi!
Is it possible to use Asterisk for web-triggered calls? For example, if
I have IAX termination from VoipJet and I want to tell Asterisk to
Hi All,
Does any one know of a way to make a three way call
from Asterisk using X-Lite.
I need the ability to be able to call someone on the
PSTN using my IAX provider then bring another person from a local extension
into the call if needs be?
I believe most three way calling is
I just downloaded the new astcc and it includes now a new
field in the list of the cards: Brand Great!
How can I use it in the dialplan?
You can't use it in the dialplan, you use it when creating a card.
Nabeel
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Hi there, we are looking for an opensource or commercial * based Call Center.
Full ACD, call monitoring, multiple queue, IVR, voicemail, management,
reporting, CDR, etc is needed. over 100 seat can be the initial target
and will grow in a very short time.
SIP phones will be used and multiple E1
Ed,
thanks for offering your help. Now it's working. My context had
a typo in it. I'd rather not talk about it ;-)
Cheers
Sascha
I just made this work without much difficulty, so I imagine you can do so
too.
It does not look like anything wrong with your queue entry.
Can you post your
I installed Asterisk on Fedora core 3, installation was successful but
when I start asterisk with vvvc after 5-10 mins the box freezes,
don`t know why, and it only happens whn I start asterisk, now i`m
installing RH 9.
fc3 is fine
u can run it in runlevel 3cli mode
Best regards,
Mohd. Imran
I appologize for the long, new-ish question, but after a few days of trying to
work a solution by reading through the list archives and WIKI and coming up
with what I thought would work, I think I'm just not getting a fine detail.
I titled this thread Setting up Security Groups because I'm
Hi
To install asterisk realtime we have to get the
asterisk from CVS, is this stable and good to use ? any bugs
?
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Jagan Mohan wrote:
I need to do load balancing only for the following functionalities:
1) Registration of SIP clients to * servers.
2) Load balancing of the INVITEs from SIP clients to different *
servers.
I'm not interested in supporting the features, which you have
mentioned below.
Hello,
We use and develop the astGUIclient suite. It is Open-source(as in GPL) and
offers Inbound and Outbound call center functions with reports, ACD,
monitoring, recording and very basic IVR scripts. Complex IVR functions need
to be custom programmed within Asterisk but that is not really that
Hi all,
I have a quick question I hoping someone can help me with. I have
[EMAIL PROTECTED] running and working just fine. I've integrated it with
BroadVoice and so far I'm blown away by everything I can do.
I don't particularly like sitting my entire machine in the DMZ on my
network sitting
Nabeel Jafferali wrote:
I just downloaded the new astcc and it includes now a new
field in the list of the cards: Brand Great!
How can I use it in the dialplan?
You can't use it in the dialplan, you use it when creating a card.
HOW do I use it?
I could not find anything useful with BRAND,
I have a quick question I hoping someone can help me with. I have
[EMAIL PROTECTED] running and working just fine. I've integrated it with
BroadVoice and so far I'm blown away by everything I can do.
I don't particularly like sitting my entire machine in the DMZ on my
network sitting
On Tue, 2005-03-15 at 07:03 -0600, Bryan Boatright wrote:
I've read several of the Wiki sections on Voicemail and Asterisk
variables but could not find an obvious answer to this question.
I would like to run a script that post-processes the voicemail after the
Voicemail application returns
Kanishka Somaratne wrote:
Hi
To install asterisk realtime we have to get the asterisk from CVS, is
this stable and good to use ? any bugs ?
yes you need to get asterisk from cvs.
no, RealTime is not part of the stable branch.
yes, the cvs version of asterisk is pretty stable.
-Matthew
I had installed it and found out only the field of inc in your account table.
For example, you have A and B brands. A's inc is 6, B's inc is 60.
When you create a user belong to A brand. It will use 6 seconds as its
includedseconds.
Best Regards
Charles
On Tue, 15 Mar 2005 23:34:38 +0800,
I would need that a user could choose between two tarriffs,
... I thought that would be great to use Brands for that.
Like I said, a brand is used when you are creating a card. Therefore,
the markup defined by the brand is applied to the card. Simple?
Nabeel
On Tue, 2005-03-15 at 07:21 -0800, PA wrote:
Right now here is how I have it structured in extensions.conf. What
am I missing? Why would a sip-basic member be able to make toll
calls?
[default]
include = sip-basic
include = sip-operator
include = sip-superuser
You probably want to
On Wed, 2005-03-16 at 02:20 +1100, Shane Dalgleish wrote:
I've put my fire proof suit on ready to ask a question ;o)
Redhat 9
Kernel 2.4.20-8
AVM Fritz PCI
after modifying the makefile defs.h, # make
Throws up a whole bunch or errors ending with..
make[1]: *** [main.o] Error 1
James Rothenberger wrote:
I am testing a call flow in which an inbound SIP call (to the Asterisk
from a PSTN connection from a SIP VoIP provider) is not answered
(nobody there and no voicemail) and the call is terminated on the PSTN
side. After the SIP CANCEL is sent to the Asterisk from the
Therefore, the pricing in the routes table is the actual cost price?
Also, I wanted to know, when you actually generate the cards $1.00 = 10 ?
Cheers,
Sahil
Quoting Nabeel Jafferali [EMAIL PROTECTED]:
I would need that a user could choose between two tarriffs,
... I thought that would
Nabeel Jafferali wrote:
I would need that a user could choose between two tarriffs,
... I thought that would be great to use Brands for that.
Like I said, a brand is used when you are creating a card. Therefore,
the markup defined by the brand is applied to the card. Simple?
Nabeel
I think
On Tue, 15 Mar 2005, Erick Perez wrote:
Hi there, we are looking for an opensource or commercial * based Call Center.
Full ACD, call monitoring, multiple queue, IVR, voicemail, management,
reporting, CDR, etc is needed. over 100 seat can be the initial target
and will grow in a very short
Well are you calling directly from your *?
Like you use a soft phone on the same machine?
If yes then canreinvite=no/yes has no meaning.
If you have an extension and you go like this:
SIPPhone - Asterisk - Broadvoice
Then is SIPPhone on a NATed IP?
Like 10.10.10.10 is the SIPPhone.
12.12.12.12
Hello
all
I have been learning
* from almost 1 month now. It looks really powerfull. I have some problem trying
to find previous post, or solutions to common problems, advice to newbies etc in
this mailing list. There is noa forum-like tool to search thru the
posts by keyworks for example.
Thx Adam, Ill try with |t and see what happens.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev
Sent: Martes, 15 de Marzo de 2005 04:08 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call
Jolio, no, I checked the wiki and didnt see that parameter
there, but I just checked show application queue and made the necessary
modifications.
Thx Guys!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of João
AmaroSent: Martes, 15 de Marzo de 2005 04:12 a.m.To:
Asterisk
On dial command yes, wtWT just in case, and it works when Im the one that
originated the call, but for example, I have the same problem that you have
when the call comes in thru a Zap channel. I cant make transfer to work
eventhough the dial command that sent the incoming Zap call to me has wtWT.
On Tue, 15 Mar 2005 11:56:18 -0500
Fabian Borot [EMAIL PROTECTED] wrote:
Hello all
I have been learning * from almost 1 month now. It looks
really powerfull. I
have some problem trying to find previous post, or
solutions to common
problems, advice to newbies etc in this mailing list.
There is
This is what you need.
Google allows you to enter a parameter called 'site:'
when you do this it searchs that site only.
The list is archived so you always have it available.
Search at google with the following...
site:lists.digium.com some
parameter
This will search the archive and you
hi folks,
I am calling from Asterisk to some Cisco gateways (as5350, 2600) and I am
having one way problems with chan_oh323.
With other provider that uses cisco also and I running the same
chan_oh323, it works perfectly.
I've tried also with chan_h323 and it does not work as well.
Asterisk cvs
Hi All,
I posted earlier with regards to three way calls and
X-Lite, this kind of yielded everything I already suspected. However I
suspect someone has a good working config for connecting a third party to an
existing call (a-la-skype), or a detailed solution of using MeetMe to achieve
On 15 Mar 2005, at 05:31, Callum McGillivray wrote:
x-tad-biggerHi,/x-tad-bigger
x-tad-bigger/x-tad-bigger
x-tad-biggerIm trying to get our TE110P card up and running on our * test server./x-tad-bigger
x-tad-bigger/x-tad-bigger
x-tad-biggerInitially we were getting a flashing red light on
1 Transcoding is between codecs. ulaw to
g.729 for example
2 I prefer AMP but unless you install it with
[EMAIL PROTECTED] it could be a pain.
3. You need a clock source for meetme and
other features to work so if you don't have any digium hardware you must use
ztdummy
4. Unless you are
Hi
I want to enable SIP calls from an ip address, direct calling without
registering, the ip which sends the calls will not change. i have the
following in the sip.conf file
[cisco4]
type=friend
host=192.168.0.5; This device registers with us
canreinvite=no ; Asterisk by default tries to
To search the list archives use this in Google:
site:digium.com search-terms
-Original Message-
From: Fabian Borot [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 15, 2005 10:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Newbie
Hello all
I have been
I've installed Asterisk from the Asterisk @home distribution.
Ultimately I will be using Asterisk for voicemail for about 150 users.
Calls are (mostly) handled by a legacy PBX although we do have a couple
of Cisco 1760 routers that connect a remote office.
I've setup a SIP trunk that routes calls
Let me rephrase this:
If yes then canreinvite=no/yes has no meaning.
Actually it does have a meaning but in this context it doesn't affect the
ability to place a call.
So:
We start from:
SIPPhone - Asterisk - SIP Provider
If canreinvite=yes then after the initial INVITE you get:
SIPPhone -
Or if google is too complex, http://asterisk.keystreams.com
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Robert Webb wrote:
On Tue, 15 Mar 2005 11:56:18 -0500
Fabian Borot [EMAIL PROTECTED] wrote:
Hello all
I have been learning * from almost 1 month now. It looks really
powerfull.
I already have qualify=yes
i will try the qualify=100
Thanks!
Randy
Brian Dingman wrote:
You get this when you lose registration. Try qualify=100 or
qualify=yes, to see if that alleviates the problem.
I can make and receive calls for about 30 seconds before this happens.
On Mon, 14 Mar 2005
In having configured my first * server there are a few questions I could not
understand or find answers to
1) How does one use ztmonitor to adjust the rxgain and txgain. I have set
mine to -1.0 each to get rid of echo on std phones connected on the TDM10B
FXS module
2) Is it best to use a TDM
Just to be safe, I removed everything and got a fresh
copy from CVS. Again I got Asterisk to compile fine,
but the addons do not. Just to double check you got
this to compile on OS X? Here is what I get.. I did
change from static to dynamic which seems to get
things further with gcc in OS X.
cc
On Tue, 2005-03-15 at 17:36 +, Barry FAWTHROP wrote:
In having configured my first * server there are a few questions I could not
understand or find answers to
1) How does one use ztmonitor to adjust the rxgain and txgain. I have set
mine to -1.0 each to get rid of echo on std phones
Title: AntiSpam Alert: Request For Authentication
Is there anyway we can get this shit off
the asterisk list apart from posting their email address [EMAIL PROTECTED] here for the spambots
to pick up?
Dean
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
MySQL: Speed, Power and Precision
_
MySQL is free. It can be installed in less than 59 minutes from source
for light use by a first time user AND there is no need for extravagant
tuning. Out of the box, it comes with 4 database engines: Isam, MyIsam,
Heap, InnoDB,
Is there any development ongoing with ICD ? I wouldn't want to get
involved in something that is not going to make it into CVS HEAD.
I really would like to use this - just worried :)
Julian
Peter Svensson wrote:
On Tue, 15 Mar 2005, Erick Perez wrote:
Hi there, we are looking for an
On Mar 15, 2005, at 19:04, dean collins wrote:
Is there anyway we can get this shit off the asterisk list apart from
posting their email address [EMAIL PROTECTED] here for the spambots
to pick up?
I believe these do not go to the list, but to people who post to the
list. You cannot turn it off
Hello!
I am new to the list and got a simple question. Is it possible to connect
with a headphone to Asterisk on the same machine?
I tried a portrange 5060:5061 in sip.conf but that doesn't work. I can
connect with sjphone to an Asterisk on a remote machine only.
Thanks in advance!
--
forum-like tool to search thru the posts by keyworks for example.
You can use google by specifying site:lists.digium.com before or after the words
Most if not all of your questions are answered on the wiki (which does
not seem to be responding as I write this) and at sites like
Anybody got a few decent pointers to get people started with ICD? We'd
like to integrate that in Xc-Ast too.
Cheers,
l.
In data Tue, 15 Mar 2005 18:12:18 +, Asterisk [EMAIL PROTECTED] ha
scritto:
Is there any development ongoing with ICD ? I wouldn't want to get
involved in something
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