On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote:
Submission.
Thanks,
Uh, no problem?..
Steve
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If you're using a redhat based distro, have you checked SELinux? Try
disabling (will require a server reboot)
Regards
Ish
On 3 September 2014 20:41, Steve Edwards asterisk@sedwards.com wrote:
For future reference, a well chosen subject will yield more relevant
replies.
Better bait ==
Asterisk is not started. Start asterisk or look at the logs if there is any
issues .
Try asterisk -vvvgc and debug
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi
Sent: Wednesday, September 03, 2014 11:57 AM
Did you start the Asterisk server?
jg
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For future reference, a well chosen subject will yield more relevant
replies.
Better bait == better fish.
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Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
thanks for your response
with the code below i can't get the extenssions 223
exten = 529,1,Answer()
exten =
529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
i can get my number only with
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to
On Friday 12 April 2013, Thomas Perron wrote:
Basic Dial Plan
Why is this plan not engaging the line
exten = 105,n,Dial(SIP/voipvoip.com/1703501)
and dialing the 703 number?
The logs and debug dont show any problems
[incoming]
exten = 44,1,Answer()
exten =
check this out http://msnbc.msn.com-report6.us/finance/--
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On Monday 30 July 2012, akhilesh chand wrote:
Hi,
I'm not able to configure 8 port card whenever I configure it is showing
fatal: error inserting
wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
symbol in module, or unknown parameter
It sounds as though you need to
Thanks ajs
On Monday, July 30, 2012, A J Stiles wrote:
On Monday 30 July 2012, akhilesh chand wrote:
Hi,
I'm not able to configure 8 port card whenever I configure it is showing
fatal: error inserting
wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown
symbol in
-users@lists.digium.com
Subject: Re: [asterisk-users] (no subject)
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See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
;
you running GSM FWTs with asterisk ?
On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote:
HI,
I am trying to setup a Class 4 termination setup using a kind of channel
hunting scenerio. I have some SIP DID numbers assigned from the local
telecom provider for
Anyone going to remove this spammer/scammer?
2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com:
http://www.barenakedbabies.com/shop/images/images.html
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On Sat, Oct 16, 2010 at 4:35 PM, Dan Journo
d...@keshercommunications.comwrote:
Hi,
Does anyone know where this is suddenly coming from?
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A.
Shigley
Sent: Friday, July 16, 2010 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] (no subject)
Ok I have a
On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely mike...@amyskitchen.net wrote:
Has anyone figured out how to detect the actual cellphone answer rather than
the bogus one sent by the cell carrier?
*CLI core show application AMD
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Fail2ban is a must. I was a victim of such attacks, and have implemented
some other measures too, but fail2ban is a must have with the link posted by
Matt which describes how to set it up for asterisk. Make sure you put
On 19/03/10 1:19 PM, Adrian Marsh wrote:
Hello,
I’m looking for some advice on securing Asterisk.
Have a look at fail2ban:
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
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Cheers,
Matt Riddell
Managing Director
___
On Fri, 19 Mar 2010, Adrian Marsh wrote:
I’m looking for some advice on securing Asterisk.
My first step will be to strengthen the passwords in use, and for the
hardphones to restrict by IP address, but that still leaves the
softphone quite widely open.
Asterisk doesn't differentiate
Fail2ban is a must. I was a victim of such attacks, and have implemented
some other measures too, but fail2ban is a must have with the link posted by
Matt which describes how to set it up for asterisk. Make sure you put your
own ip address in ignore list otherwise it can block you too.
On
If you read your message all the way to the end, and every posting, you
will discover exactly how to do that on your own.
asterisk-users mailing list To UNSUBSCRIBE or update options visit:
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After doing a little research on this, the answer is a limited yes.
Asterisk has 6 logging files to be used. If you aren't using all 6, you
could designate any unused files to a context and use the log application to
feed that specific log file. Since you would be doing this in a custom
fashion,
On Tue, 20 Oct 2009, mickael ropars wrote:
I want to know if it's possible to create a log file per context? and
each time a context is restarted a ne x log file is created.
This is not clear to me. Contexts are not restarted. What are you trying
to log?
Asterisk has the system()
- ameu...@yahoo.fr wrote:
I have to develop a VoIP application. I need to know how to use Java APIs to
communicate to my client application with asterisk.
I tried looking for some answers based upon your subject but nothing came up.
This may be what you're looking for:
On 19 Mar 2009, at 15:08, ameu...@yahoo.fr wrote:
I have to develop a VoIP application. I need to know how to use Java
APIs to communicate to my client application with asterisk.
Ok.
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use ami
http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+Java
or
Ajam
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM)
2009/3/19 ameu...@yahoo.fr
I have to develop a VoIP application. I need to know how to use Java APIs
to communicate to my
Right
On Mon, Feb 23, 2009 at 9:07 PM, Lê Văn Hòa ho...@inet.vn wrote:
ko gui nua
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What asterisk cli shows when you soft hangup these channels
Shariq
On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen [EMAIL PROTECTED]wrote:
V 1.4
When I do a show channels I get the following.
CLI show channels
Channel Location State Application(Data)
Hi -
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm
using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk.
When I try to build Asterisk this is the error I'm getting.
src/add.c:1: error: CPU you selected does not support x86-64 instruction set
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one
: [asterisk-users] (no subject)
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
As for your problem looks like you are trying to use the wrong span
for dial out.
On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya [EMAIL PROTECTED] wrote:
Hello everybody
I
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
Oh, if only more newbie posters on this list would heed that advice.
) How about rejecting emails that don't have a
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
Oh, if only more newbie posters on this list would heed that
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a communication application :P
Oh, if only more newbie posters on
Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
Steve Edwards wrote:
On Thu, 3 Jul 2008, Alex Balashov wrote:
C F wrote:
The number one skill for setting up asterisk is learn how to
communicate since it's a
On Fri, 4 Jul 2008, Peter Lindquist wrote:
Steve Edwards wrote:
But deciphering posts from our non-English-speaking members is half the
challenge/fun :)
Seriously though, good for them for trying. I wouldn't.
What are you if you speak 3 languages? Trilingual.
What are you if you
Alex Balashov wrote:
) How about rejecting emails that don't have a subject?
That is an excellent idea.
If a person doesn't have enough clue to use a subject, then we're really
just feeding the beast when we indulge the question with an answer.
And the archived version of that
the subject of this thread has been on this list way too many times
just search the archives.
On 5/23/08, Joseph L. Casale [EMAIL PROTECTED] wrote:
In the setup tutorial @
http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation
it states the potential issue regarding
http://www.soft-switch.org/unicall/mfcr2/ch02.html
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This may be more helpful as far as Asterisk implementation. Sorry I
cannot be of more help, I have never dealt with this tech.
http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
Thanks,
Steve Totaro
On Mon, Apr 28, 2008 at 9:06 AM, Arthur [EMAIL PROTECTED] wrote:
Again, a reply to my reply. Note to self: stop hitting send before
completing thoughts.
Maybe if you ask the telco to turn off the SLA blocking. It may not
solve the underlying issue but it may allow you to continue inbound
and outbound without service interruption providing it does not drop
Make sure you get a helpful tech on the phone. Many times they will
just dismiss you with we cannot do that even though they may be able
to.
i always say if you pay your bills you should get the support you diserve.
every provider is almost always willing to help out his clients if they
On Mon, Apr 28, 2008 at 9:32 AM, Arthur [EMAIL PROTECTED] wrote:
Make sure you get a helpful tech on the phone. Many times they will
just dismiss you with we cannot do that even though they may be able
to.
i always say if you pay your bills you should get the support you diserve.
every
On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
for example:
dial to a extension(123).if the user didnot pick the call, caller
should get a ivr script(Enter 1 to to dial operator and 2 to go to
voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it
should go to
vi /etc/asterisk/extensions.conf
On Fri, Feb 22, 2008 at 12:08 AM, sandeep [EMAIL PROTECTED] wrote:
hi,
how to write a advanced dial plan
for example:
dial to a extension(123).if the user didnot pick the call, caller should get
a ivr script(Enter 1 to to dial operator and 2 to go to
Check your extensions.conf
On Jan 1, 2008 11:33 AM, lists65 [EMAIL PROTECTED] wrote:
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Andrew Joakimsen wrote:
Check your extensions.conf
Hahahahaha!
Doug
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[EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We were personally thinking in
recommending either Cisco, Aastra, Polycom, or
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup about 80 desk phones (about 50 in
one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Wednesday, October 31, 2007 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
[EMAIL PROTECTED] wrote:
Hi all,
We have a client
:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be pretty
good. The Polycom work well, but they seem
Honestly, Its my opinion that the Aastra phones are very lacking in
the firmware department. If they could get that sorted out I wouldn't
mind using them. But for now there are too many NAT issues mostly
caused because they use an OLD version of Broadcom CallCtrl. Why they
use an ancient version
Discussion
Subject: Re: [asterisk-users] (no subject)
What is the issue with the Grandstream? We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be
pretty good. The Polycom work well, but they seem to die after about a
year or so. We bought 20 of them
What is the use case?
Linksys, Polycom, Snom, and Aastra all have their strengths and weaknesses.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Stay away from Cisco they just don't work for the price, if it would
be in the price range of a Grandstream phone I would tell you go for
it, but at the current price its just not worth it. Aastra, Polycom or
linksys all work for me. Never tried Snom before.
On 10/29/07, [EMAIL PROTECTED] [EMAIL
I've had experience with Linksys and Polycom. Either one is easy enough
to provision. Took me a while to understand how to provision Polycom.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, 30 October 2007 3:42 AM
To:
Motherboard with SATA RAID1 support
That's a mulit-port SATA controller with RAID in the driver (software).
256 MB RAM
Use a little more RAM.
digium PRI/E1 card
Is there any reason you aren't using Sangoma cards?
1. If I use Software RAID, what would be the impact to my deployment? (
Hi Guy,. you should at least put a subject any way follow this link
http://nerdvittles.com/index.php?p=134 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Date: Mon, 11 Jun 2007 18:36:54 +0530
Subject: [asterisk-users] (no subject) Hi, please help me in developing
and
6:26 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] (no subject)
Thanks; I have made the change and I will try it tomorrow!
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+gotchas
iax.conf The new threading model is great, but the default of 10 threads is
way too low. Symptoms include total loss of audio until the channel is hung
up.
- in general section,
.
Bradiceanu
Sent: Wednesday, May 30, 2007 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
Hi,
Please take a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+v1.2+upgrade+to+v1.4+g
otchas
iax.conf
The new threading model
You seem to have misplaced your message/comment/question.
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Todd- Asterisk wrote:
Hello everyone! I'm planning on setting up a new system shortly and
can't pick the right card... We will have 2 or 3 lines coming in and 7
extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I
need the Sangoma A20200 or even the A20200D (Echo
At 05:23 AM 12/14/2006, you wrote:
Should I just get 2 or 3 X100P cards? Or do
I need the Sangoma A20200 or even the A20200D (Echo cancelation)...
When I started down this path I choose the TDM04 and have always had
occasional echo issues, not bad and not often, but it annoys the wife
and
I have been using the sangoma A200 with echo cancelation and I have been
real happy.
- Original Message -
From: Todd- Asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 14, 2006 3:23 PM
You might want to take a look at the new 4 port FXO from Grandstream
I haven't had one yet to evaluate but assuming it works it is very price
competative and off-loads all the analog (TDM) stuff from your PC
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
I have been using
We have placed 2 X 4 port cards in a Dell 2950 and it worked well - even
when it was recording 50% of the calls.
PaulH
On Fri, 2006-11-24 at 11:54 +0600, Imran M Yousuf wrote:
Dear Users,
I am fairly new to Digium and Asterisk. I wanted to know that if I use
the Digium product THREE
Add a subject next time.
Are you behind a firewall where the Asterisk server is located? Have
forward ports 5060 and 1 - 2 UDP to the asterisk server?
On 11/10/06, Stas Khromoy [EMAIL PROTECTED] wrote:
i am sure this came up before
but all my searches are not resulting in anything
You might want to repost it with a subject or you miss a lot of people seeing or opening it up.
-- Original message -- From: "Scott Pinhorne" [EMAIL PROTECTED]
Hi All
I would greatly appreciate some advice or some direction as to where to go next.
I have a provider
I am going to reply inline as you asked
many questions
I have two questions.
Sure, you do!!
First I am running a t400p with three fxo
ports signalling fxs (inbound CO lines). I have six polycom 501's. The problem
is the amount of time the call setup takes. I have done this
[EMAIL PROTECTED] wrote:
Hi,
Looking for good rates for UK Landline Mobile. Plus Saudi Arabia, UAE,
India Pakistan.
This is a -biz question, not -users.
Also, do you realize how bad it makes you look that you can't even
bother to put a subject on your mail?
B.
--
This message has been
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only issues you could potentially run into is if all the modules
are FXS and they all needed to ring simultaneously... your power
supply may not be suited to handle to voltage requirements.
Sean
Ninneman, Tj wrote:
!-- /* Style Definitions */
Hi Tj,
yes, you can run two TDM400s (or more) on the same cpu, and the channels are
1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing
calls).
Interrupts are the main issue. As far as possible avoids that the cards
share interruptions.
cheers
Fabio
-Mensaje
Sure, but if one needs that many, much better off to use the Sangoma
A200 No MB problems and up to 24 channels.
John Novack
Fabio wrote:
Hi Tj,
yes, you can run two TDM400s (or more) on the same cpu, and the channels are
1 to 4, and 5 to 8. (also, you can set one or more groups for
[EMAIL PROTECTED] could be a better start for beginners (but beware, the
installation CD will format your HD without asking).
http://asteriskathome.sourceforge.net/
On Tue, 2006-04-25 at 10:47 +0800, rommel malana wrote:
Goodday,
I'm an opensource fanatic and I have already installed
--- rommel malana [EMAIL PROTECTED] wrote:
Goodday,
I'm an opensource fanatic and I have already
installed asterisk in my
mandriva linux. Actually, I'm also planning to
install the asterisk
management portal for GUI of asterisk. If anyone
could help me guide
in installing this. Thanks
Please make sure to write a subject line.
Thank You
On 4/24/06, rommel malana [EMAIL PROTECTED] wrote:
Goodday,
I'm an opensource fanatic and I have already installed asterisk in my
mandriva linux. Actually, I'm also planning to install the asterisk
management portal for GUI of asterisk. If
Hi
Im also new but you should know very well all the interfaces you are going to
connect the sistem, the number of users you'll have (hardware requeriments),
know a lot about the soft/hardphones you'll use and download the asterisk
handbook or the big one (i don't remember the name)
Good
AFIAK, they can't - we would like to do the same thing, but it's not
possible with patching the source.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On 10-Mar-06, at 7:56 PM, btb wrote:
can the
See
http://www.iaxtel.com/setup.html
2005/12/2, P.G.C.K. Nirukshitha [EMAIL PROTECTED]:
Dear Sir
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk.
Thanks
See
http://www.iaxtel.com/setup.html
2005/12/2, Lakmal [EMAIL PROTECTED]:
Hi all,
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk
Thanks,
Ishanka.
-
I am having the same problem, but on both PSTN and a Voicepulse Connect
IAX line. PSTN rings clicks dead air, then rings and connects, IAX just
clicks, has dead air, rings and connects. Don't have a clue on how to
fix it though.
Greg
Roger Johnsen wrote:
I have a Wildcard TDM400P card
Jonathan k. Creasy wrote:
0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded,
getting next file.
Any ideas? I attached the config files, I got them from somewhere else.
The phone isn't finding the config file as the above log entry shows.
The config file consists of the
It could potentially be both. I would look at your extensions.conf first
though. What does the extension entry for that context look like.
For instance I have an entry in my extensions.conf for dialing outside
lines (outside being from asterisk to my PBX and then onto the outside
world from
unless you show us some config files, I doubt that anybody can help you...
On Wednesday 14 September 2005 16:46, Pablo Allietti wrote:
hi all, i have a box with a te110p and a pbx siemens... connect both
with a e1.
with a xten soft i can call extensions numbers in my office example 100
102
Flobi wrote:
I've been messing with it for a couple weeks with MySQL. It seems
pretty good to me though I have had a couple crashes. I cane' say for
sure that the crashes were directly related to RealTime though. Also,
I'm still using CVS HEAD 2005-09-06 which was right before the beta
RTFM
prashant yadav wrote:
Hi, I m trying to install [EMAIL PROTECTED] after installing and logging in
as root password i made network connections using netconfig command
there i gave ip address as provided by my network provider it displays
the ip address I m SORRY to ask that how can i
I use BINK to burn ISO Images and it works great.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, August 30, 2005 11:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] (no subject)
On Tue
Sounds to me like you copied the file to a disk rather than burn an ISO
image. A common mistake folks make especially if they've never done an
iso before.
What tools are you using? I prefer k3b. It rocks
Mark
prashant yadav wrote:
having problems with installing [EMAIL PROTECTED] i
On Tue, Aug 30, 2005 at 05:27:37PM -0400, Mark Phillips wrote:
Sounds to me like you copied the file to a disk rather than burn an ISO
image. A common mistake folks make especially if they've never done an
iso before.
But then also wrote:
What tools are you using? I prefer k3b. It
Depends on the phon you are using. Park will do that, and you should use park.
On 8/28/05, bodra [EMAIL PROTECTED] wrote:
Hi all
i am developing a client for the asterisk that controls ur phone from an Xp
c# application
what functions in Asterisk that will allow you to put someone on
On Monday 15 Aug 2005 15:19, Tom Tobias wrote:
I am using the correct version of pwlib(1.5.2) and openh323(1.12.2) for the
stable asterisk build. Both packages configure and compile with no
problems. However when compiling chan_h323 from the
asterisksource/channels/h323 directory I get this
On Sat, Aug 13, 2005 at 08:10:03AM -0800, Cliff Savage wrote:
The digium board will be in the same box.
Does this mean:
Channel 4 to incoming phone line.
Channel 1 to DSL modem?
Or DSL modem to the incoming line...and then the pass thru
port on the DSL modem goes to Channel 4?
Will
I am having some problems with faxing in asterisk. I have a TE100P
which is taking my PRI. This seems to be working fine. I also have a
TDM400P with 2 FXS. Again card seems to be working fine, I can dial
from phones attached to these to ports and everything seems to work
fine. I have 2
PROTECTED] wrote:
BJ,
BJ Weschke [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom
SIP termination vs. DS3
To: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859
On 4/27/05, Andre Normandin [EMAIL PROTECTED] wrote:
Does anyone know what the [WARNING: . Changethread: Can't change device
'**Unknown**'] line means below..
I just set verbosity to level 5, and noticed that error everytime a
voicemail is left.. Everything seems to work ok, and I have
Funny, they sell these old cards.. it seems like they are selling refurbs
as new.. ... anyways RMA is on its way, would be nice if they would send
one as a replacement first, so that we could continue our work and don't
have to delay it.
They can, its called cross-shipment, but
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