Testing once again.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
testing
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
testing yet again.
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Please disregard.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi
why my posting are not accepting in this list
ram
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or upd
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Test from non-digium email.
--
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the
This is just a test of the asterisk-users mailing list.
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.ap
Is there any way to "prove" that an EnumLookup is actually being done?
I've been trying to get ENUM working, and have gotten to the point where
I'm pretty sure the NAPTR records are resolving the way they ought to
be, and "manual" lookups using dig return just what they "ought" to.
But asterisk
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
So, now that we've all complained about the state of testing of Open
Source versions of Asterisk, lets do something about it.
I propose we start with a list of things that we think should be tested
in Asterisk, and means to test them.
Maybe we c
While I read on some other mailing list that the human ear is a poor
testing device, it is still a widely available testing device and I
often don't have anything better.
In order to help that device better detect sound quality issues, I tend
to prefer to use lengthy music files. Once I'm familiar
Hello
Can or is there somewhere a way to test my outgoing H323
I like to connect to a terminating server but I'm still getting hangups.
Phone is ringing on the othersite but my asterisk telling my no one
availble at this moment.
Like to test my H323 loutgoing line.
I't looks so stuppid if someth
It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.
How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?
Is there some circumstances where you can ask permission from the city
ahead of time?
I realize
Hello gurus,
After successful installation of Areski. I am
having few problem before I can do any test dial-outs.
When I try to create sip/iax
friend from web interface it says"Could not open buddy file
'/etc/asterisk/additional_areskicc_sip.conf'
I tried creating the file manually with
ram wrote:
> Hi
>
> why my posting are not accepting in this list
Don't know.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineap
If I see this, then messages are getting through.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici
[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api
I'm having problems sending e-mails to the list. Please ignore this
message, I'm just testing. sorry for the inconvenience.
Thiago
Abra sua conta no Yahoo! Mail, o único sem limite de espaço para
armazenamento!
http://br.mail.yahoo.com/
___
-
Matt Riddell wrote:
> Should these tests be added to Asterisk-Addons or maintained outside of
> the tree?
If people start writing test utilities, I would be happy to host them in a
subversion repository. Depending on the size of this stuff, it could probably
go into the main Asterisk repository.
On 8/30/07, Russell Bryant <[EMAIL PROTECTED]> wrote:
> Matt Riddell wrote:
> > Should these tests be added to Asterisk-Addons or maintained outside of
> > the tree?
>
> If people start writing test utilities, I would be happy to host them in a
> subversion repository. Depending on the size of thi
matt,
are you looking for unit testing of the * components or systems testing,
testing the finished product? or both?
I think you are onto something here... I hope it takes root. I would
say put it in the addons. it would be Great if digium takes it up. it
is a smart move for them to foster,
I think the testing frame work includes both the components
and system testing.
I wish to add some more test even though all giants may aware,
since i wish to do some contribution to asterisk what ever i can.
i am plannig for the framework and addon as given below, expecting
techies advise in this
Matt Riddell <[EMAIL PROTECTED]> writes:
> Hi,
>
> I propose we start with a list of things that we think should be tested
> in Asterisk, and means to test them.
>
> Any takers? Add to the list? If there is something you believe is
> mission critical to your business, write up a test case for it
On 11 Aug 2009, at 12:03, Tzafrir Cohen wrote:
> So I'm looking for a music file that:
>
> 1. Sounds well (enough) even at the standard PSTN quality (8kHz, mono,
> 16 bits per sample).
>
> 2. Is long enough. E.g. ~10 minutes.
>
> 3. No usage limitation. Freely usable. So I can point it out to
>
All,
I'm trying to troubleshoot why I can't dial out using X-Lite via *. I
have a TDM01A (1 FXO), latest *, and latest AMP. * starts successfully,
and ztcfg -v shows that 1 channel is configured. However, when trying
to dial out I get a "all circuits busy" message.
When at the CLI, "show chann
(Sorry if this is a repost...I submitted this a while ago, but it never
showed up on the list)
All,
I'm trying to troubleshoot why I can't dial out using X-Lite via *. I
have a TDM01A (1 FXO), latest *, and latest AMP. * starts successfully,
and ztcfg -v shows that 1 channel is configured. How
I need to test QoS on an IAX link between a server in Colorado and a
server in Europe. I know I could install a Milliwatt extension on the
European server and just listen, but is there a more scientific method
to collect QoS metrics?
Thanks
P.S. I'm getting a lot of "Page Not Found" on list
I don't think it's a stupid question at all. Testing 911 routing is
very important, and it would suck to find out it didn't work when you
needed it to. When I tested 911 at my wife's small business (we're
on ZAP channels), I first called the non-emergency number for our
local police depar
On Jul 16, 2006, at 11:05 PM, voiplist wrote:
It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.
How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?
Is there some circumstances where you can ask pe
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voiplist
Sent: Monday, July 17, 2006 2:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing 911?
It seems that 911 is important enough that when setting
bject: RE: [asterisk-users] Testing 911?
I call and immediately identify this as a test call.
I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them. If all is OK I thank them and hang
voiplist wrote:
It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.
How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?
Is there some circumstances where you can ask permission from the city
ahead of
I do it all the time, after I finish installing a PBX (asterisk or
other PBX) I dial 911 and say: Hi this is a test call, I'm a PBX tech,
just finished an installation and just wanted to make sure that 911
works. Then I ask the operator on the other end of the line to confirm
the e911 info he has
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, July 17, 2006 5:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing 911?
voiplist wrote:
> It seems that 911 is important enough that when setting up an Asterisk
> box, i
Not an answer to your questions, but just in case you don't know there
is a lot of info on the wiki:
http://www.voip-info.org/wiki/view/AreskiCC+CallingCard+Application
We use Areskicc here, and it works great. However we do not use sip/iax
friends, perhaps both of your problems lie there?
B
> When I try to create sip/iax friend from web interface it says
> "Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf'
> I tried creating the file manually without luck.
Make sure the user your web-server runs as can write to and read from the file.
>
> Second I am unable t
Hi there, im testing my asterisk box using a Modem
Intel 56K which on the documentation says it must have
the same behavior as an X101P. So im trying to
configure just a simple line with 6 extensions.
Asterisk loads fine and when im testing an incoming
call the welcome message answers but when im t
Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live
production.
I cannot run valgrind and I do not have the right flags set in menuselect.
I can however at the dead of the night run stress tests.
I want to simulate x-amount of concurrent calls to both a dtmf dia
I'm in the process of setting up an asterisk box that will stand
between PBX's and the SIP providers. So a trunking server.
How can I 'test' to see if this trunking server is stepping out of the
media path during calls?
Thanks
David
--
--
www.ringfree.biz
828-575-0030
--
__
How to test 911 call?
I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call.
I don't want to go live as they might charge me.
--
Joseph
--
_
-- Bandwidth and Colocation Provided by ht
testing users list
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asteris
I have just posted my initial impressions of using asterisk with
broadvoice.com, available at
http://www.dslreports.com/forum/remark,9764095~mode=flat
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-us
On 04/08/08 07:23, John covici wrote:
>If I see this, then messages are getting through.
>
You are lucky :-/
I sent two messages to the list and they never arrived.
--
#Joseph
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
thanks
Hi all,
I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our
asterisk installation. We have a very simple dialplan that uses FastAgi.
I'm finding that all calls to "GET VARIABLE" from the FastAgi are
returning null when the dialplan is invoked from sipp -- and they work
fi
Hello,
Does anyone know how to test the timing device?
I've tried the following but with no luck.
Zaptel is installed.
I'm trying to use ztdummy as a timer.
[r...@templateasteriskserver ~]# dahdi_test
Unable to open dahdi interface: No such file or directory
[r...@templateasteriskserv
Hello Asterisk-Users,
My organization is putting together a VoIP setup with Asterisk for a
call center, and we currently have two identical machines for
redundancy. How do you test your redundant machines? How do you test for
load problems? Do you have a strategy or regular plan?
I'm most concern
> When at the CLI, "show channels" shows nothing.
Look for ztcfg
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/ma
Dear all
i am new to asterisk i dont have any asterisk supported modem yet
i want to test asterisk for voip calls only
will firefly softphone will work on it ?
what configuration needed to recieve voip calls on asterisk
Thanks and Regards
Talha
Hello,
I'd like to be able to see if a channel is use and handle the call
differently if it is. The best I can find is the command
ChanIsAvail(). The problem is, I have an snom200 phone which does call
waiting, so even if it is engaged in a call, a second channel is still
available on it. I
A Chairde,
I am looking to test my Asterisk server with PROTOS,
but having some problems.
I am trying commands.The PROTOS tool definitively
works, I think I am putting wrong TOURI in ( [EMAIL PROTECTED] ) ,
123456 is number of softphone hanging off switch, not th
A Chairde,
I am looking to test my Asterisk server with PROTOS,
but having some problems.
I am trying commands.The PROTOS tool definitively
works, I think I am putting wrong TOURI in ( [EMAIL PROTECTED] ) ,
123456 is number of softphone hanging off switch, not the
FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message
every 20 sec.
# Testing 10.0.0.203 with 10.0.0.0
10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in
sip.conf. Asterisk server is on 10.0.0.26 address.
Why do I get this message?
sip.conf
[general]
ex
:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Testing with X101P
Hi there, im testing my asterisk box using a Modem
Intel 56K which on the documentation says it must have
the same behavior as an X101P. So im trying to
configure just a simple line with 6 extensions.
Asterisk lo
Hi,(I tried to post this message a week ago but I don't think it could reach the list. Please forgive me if you already received it).I would like to develop my first FastAGI script.I would like to test
it independently from Asterisk for the sake of simplicity.Which linux
(or cygwin) tool is the b
I have asterisk with rxfax txfax modules.I want
to test fax sendig and reciving in one asterisk
instance, in extensions.conf I have :
exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug)
exten => s,1,Dial(1234567)
exten => s,2,txfax(/home/patryk/fax.tif|caller|debug)
but I doesn't seem to w
A little look at the dialplan which rings your extension, or get dtmf, and
plays DTMF will help better understand. btw you can set the
context/extension/priority in a call file to skip some priorities of a
particular extension set.
On Fri, Sep 16, 2011 at 12:18 AM, ERIC HERRON wrote:
> ** **
>
Am 15.09.2011 21:18, schrieb ERIC HERRON:
>
>
> Asterisk 1.4.26 keeps randomly crashing then restarting itself on my
> live production.
>
>
>
> I cannot run valgrind and I do not have the right flags set in menuselect.
>
>
>
> I can however at the dead of the night run stress tests.
>
>
tcpdump and wireshark would help I guess. Just sniff for sip traffic and look
out for what's happening there. My 2 cents
Sent from my iPhone
On May 19, 2012, at 8:33 PM, David Wessell wrote:
> I'm in the process of setting up an asterisk box that will stand
> between PBX's and the SIP provider
Joseph,
I have made a quite a few test calls to 911. They don't charge you and they
don't get upset.
Just let them know right away it is a non-emergency test call, and then let
them know who you are and what you need to verify on their information screen.
Mark Engelhardt
On May 5, 2013, a
If there is a non-emergency number you can call and let them know you would
like to do some test calls. This also allows you to schedule a time for
testing when the PSAP is not as busy allowing for real calls to be handled.
On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt <
ma...@intuitiveenginee
I actually work in a 911 center. Please do not dial blindly to do a test
call. Please call the non-emergency dispatch number, ask if it would be ok
to make one or two test calls. If they give you the ok, please complete
those calls as quickly as possible as conditions change in an instant. If
t
...@gmail.com>>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Date: Sunday, May 5, 2013 8:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [asterisk-users
Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.
On 1/31/07, fadi mujahid <[EMAIL PROTECTED]> wrote:
Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application i
Thanks for ur suggestion.
But the problem is that won't test the queuing of the outbound and inbound
calls of the callcenter
thanks again
On 1/31/07, Alejandro Lengua <[EMAIL PROTECTED]> wrote:
Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.
On
:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing IVR / Callcenter applications
Thanks for ur suggestion.
But the problem is that won't test the queuing of the outbound and inbound
calls of the callcenter
thanks again
On 1/
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
Just use an
Bill
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *fadi
mujahid
*Sent:* Wednesday, January 31, 2007 10:34 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Testing IVR /
Hello,
We usually use a crossover T1/E1 cable and a multi-port T1/E1 card and
call the server from itself or another Asterisk server. We have used
this method to do stress testing in VICIDIAL, which has a builtin set
of tools for stress testing outbound dialing.
MATT---
On 1/31/07, fadi mujahid
you can use a SIP based phone service to try it out
>
> Hello
> We are developing an application to be deployed on E1 lines (inbound and
> outbound calls)
> What is the best way to fully test the application if we do not have E1
> lines in the development environment?
> Is there some kind of s
Hi,
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??
I jus
Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing the Timing Device
Hello,
Does anyone know how to test the timing device?
I've tried the following but with no luck.
Zaptel is installed.
I'm trying to use ztdummy as a timer.
[r...@templateaste
Discussion'
Subject: Re: [asterisk-users] Testing the Timing Device
You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to
load the devices or dummy devices
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun..
ourno
Sent: Thursday, October 15, 2009 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing the Timing Device
Hi Danny,
I've tried that but I get the following errors:-
[r...@templateasteriskserver ~]# /etc/init.d/dahdi start
Loading
isk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 15 October 2009 20:40
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Testing the Timing Device
What does /etc/dahdi/modules look like? I suspect that it has each of
the w
risk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing the Timing Device
Ok, its a little better now.
But I still get a fatal message:-
[r...@templateasteriskserver dahdi]# /etc/init.d/dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not f
On Thu, Oct 15, 2009 at 08:42:22PM +0100, Dan Journo wrote:
> Hi Danny,
>
>
>
> I've tried that but I get the following errors:-
>
>
>
> [r...@templateasteriskserver ~]# /etc/init.d/dahdi start
>
> Loading DAHDI hardware modules:
>
> FATAL: Module dahdi not found.
The dahdi kernel modul
I am configuring an asterisk server and I want to test the incoming
configuration with my FXS handsets.
I have the FXS lines able to call eachother and they can connect out
the FXO lines.
I changed the context for the FXS lines to "incoming" so that they
would be able to test the setup for inc
gt;
> dtmfmode=inband only works with the ulaw and alaw codecs. If you use
> any other codec you MUST use rfc2833 or info DTMF modes (set on the
> phone AND on Asterisk)
>
> --__--__--
>
> Message: 2
> From: "Jay Milk" <[EMAIL PROTECTED]>
> To: <[EMAIL
How could I test if my ports 1 to 2 are open on my firewall?
--
#Joseph
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/
At 5:07 PM -0600 on 8/20/04, Luke Cyca wrote:
Hello,
I'd like to be able to see if a channel is use and handle the call
differently if it is. The best I can find is the command
ChanIsAvail(). The problem is, I have an snom200 phone which does
call waiting, so even if it is engaged in a call, a
Sorry, I haven't received a message in a few hours, just testing to see if
it is alive.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/list
Bonjour,
En attendant la naturalisation Espagnole, es tu rentré dans les arcanes
des DEADAGI ? (qui dit Fastagi dit un peu.)
J'avoue ne pas etre sur des limites ou non de ces bestioles.
-Juste une AGI qui ne s'interromp pas sur hangup ?
-une AGI qui permet d'accèder aux $var d'un channe
I was playing with the fax stuff over IP on Friday. Unless you're
receiving faxes from a PSTN circuit, it doesn't work so well.
Also, I don't think you can chain txfax and rxfax like that. When you
hit the s,2 part, it's going to play the fax out to the handset you
dialed from. You'll need somethi
You could always use System() to copy a call spool file to launch the
outbound fax call. I don't really think a 3rd party app is necessary.
-Corey
On Mon, 27 Mar 2006, Gary Richardson wrote:
> I was playing with the fax stuff over IP on Friday. Unless you're
> receiving faxes from a PSTN ci
"You could always use System() to copy a call spool file to launch the
outbound fax call. I don't really think a 3rd party app is necessary."
Could You explain this please? Or maybe some links to
documentation and examples ?
Thanks Patryk.
___
-
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:
This is the command I send at SIPp server:
./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
This is the result I see:
Last Error: Aborting call on unexpect
Hi!
My customer want's to allow calls to landlines in EU and US and disallow
calls to cells in EU. Rest of countries are blocked.
Country blocking is easy... Is there a service that allows checking
phone number? Maybe some specific Enum? I ask for number and server
responds with info, for ex
can i test my asterisk11 on a single machine on which asterisk is installed
that sounds are working from both end properly.
i have installed asterisk 11 on ubuntu12.04 with twinkle soft phone.
regards
abhi
--
_
-- Bandwidth and Co
I'm looking to find a test tool that will register with our Asterisk
(Trixbox) server here at work and place an outgoing call via our main SIP
trunk (BroadVoice) to confirm that things are working. I've looked around
but I can't seem to find any tools that will do what I'm looking for.
I can't jus
On May 24, 2004, at 4:00 PM, Michael George wrote:
I am configuring an asterisk server and I want to test the incoming
configuration with my FXS handsets.
I have the FXS lines able to call eachother and they can connect out
the FXO lines.
I changed the context for the FXS lines to "incoming" so
s using a Sipura FXS port to make it ring. I'd still like that
$50 though :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
George
Sent: Monday, May 24, 2004 3:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] testing asterisk on FXS
r $49.99+S&H I can sell you an FXO/FXS test-cable... just kidding.
> > Use a regular RJ11 cable to connect one of your FXS ports to the FXO
> > port you want to test, pick up another FXS and dial the extension... and
> > you're promptly delivered to the [incomi
EMAIL PROTECTED] On Behalf Of Michael
George
Sent: Monday, May 24, 2004 3:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] testing asterisk on FXS lines
On May 24, 2004, at 4:00 PM, Michael George wrote:
I am configuring an asterisk server and I want to test the incoming
configuration with my F
the ulaw and alaw codecs. If you use
any other codec you MUST use rfc2833 or info DTMF modes (set on the
phone AND on Asterisk)
--__--__--
Message: 2
From: "Jay Milk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] testing asterisk on FXS lines
Date:
o peer call. I have set
dtmfmode=inband. Is there anything else I need to set?
dtmfmode=inband only works with the ulaw and alaw codecs. If you use
any other codec you MUST use rfc2833 or info DTMF modes (set on the
phone AND on Asterisk)
--__--__--
Message: 2
From: "Jay Milk" <[EMAIL PROT
Hiyall.
Just wondering how people test your emergency dialing in the UK.
Obviously you need to dial the 999 for emergency services, but am a bit
unsure if this would go down too well with the operator with a 'sorry
just testing' call. (you do all /test/ your emergency dialing dont
you!?:-) )
A
On Fri, 22 Oct 2004 13:10:38 -0600, Joseph <[EMAIL PROTECTED]> wrote:
> How could I test if my ports 1 to 2 are open on my firewall?
nmap is your friend
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages u
Benjamin on Asterisk Mailing Lists wrote:
On Fri, 22 Oct 2004 13:10:38 -0600, Joseph <[EMAIL PROTECTED]> wrote:
How could I test if my ports 1 to 2 are open on my firewall?
nmap is your friend
Or alternatively (if you don't have an outside machine to test from) you
could go to http://www.a
OK, I'm going nuts here trying to correctly identify null values,
specifically when callerID info is not available.
FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a
(debian Sid).
A snippit of my dialplan looks like this:
exten => s,1,SetCIDNum(${CALLERIDNUM})
exten => s,2,No
1 - 100 of 159 matches
Mail list logo
