my lsdahdi output is;
1. [r...@elastix ~]# lsdahdi
### Span 1: WCTDM/8 YSTDM8xx REV E Board 9 (MASTER)
1 FXSFXOKS (In use)
2 FXSFXOKS (In use)
3 EMPTY
4 FXSFXOKS (In use)
5 FXOFXSKS (In use) RED
6 FXO
On Mon, Oct 26, 2009 at 11:20:07PM -0700, PATRICK KANGETHE wrote:
my lsdahdi output is;
1. [r...@elastix ~]# lsdahdi
### Span 1: WCTDM/8 YSTDM8xx REV E Board 9 (MASTER)
1 FXSFXOKS (In use)
2 FXSFXOKS (In use)
3 EMPTY
4 FXS
On Mon, Oct 26, 2009 at 09:02:10PM -0300, Mariano Lecuona wrote:
For some reason I am not able to set loopstart instead of kewlstart:
Console out put:
[Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26
20:58:40] Found
[Oct 26 20:58:40] -- Registered channel 1, FXS
hi ,
i have started reading asterisk book need your guidance.friend as
i am newbie in asterisk so plz plz forgive me if i ask stupid questions.
INSTALLING ASTERISK
- on which linux flavour i should start the
installation of asterisk (CentOs,Fedora,Ubuntu)
right now i am using
ubuntu
- on which linux flavour i should start the installation of asterisk
(CentOs,Fedora,Ubuntu)
right now i am using ubuntu 8.04lts
I use Centos 5.3
- What are the essential packages.
The book explains which packages you need to get a basic Asterisk up and
running.
- whats is PRI, BRI
installing asterisk___
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On 27 Oct 2009, at 09:49, aster...@opensourcesolution.in
aster...@opensourcesolution.in
wrote:
installing asterisk
Me too!
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aster...@opensourcesolution.in wrote:
installing asterisk
I am intrigued by your ideas and would like to subscribe to your
quarterly newsletter, as well as attend your biannual leadership seminar.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel :
Hi,
Does anybody know when the M9 is actually being launched ? All I have read is
late October.
Best Regards,
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SplatNIX IT Services :: Innovation through collaboration
Hi All,
Could somebody explain me how the timestamps are computed in asterisk
while bridging two sip channels ?
I've got situation with my provider, who changed some things in config
and added some codecs (that much i know) and after that we got one way
audio issues. It seems that the problem
Go to sites like digium.com, asterisk.org, asteriskguru.com,
trixbox.com,elastix.org for more understanding.
Goodluck.
From: aster...@opensourcesolution.in aster...@opensourcesolution.in
To: asterisk-users@lists.digium.com
Sent: Tue, October 27, 2009 12:34:18
Hi all,
I have problem with one of my configuration :
FAX - AEX410P (One FXS port) --- BN4S0 PSTN
Case 1 : Receiving Fax is Ok ( PSTN --- BN4SO -- AEX410P -- FAX )
Case 2 : Sending Fax is nok ( FAX --- AEX410P -- BN4SO -- PSTN )
I think we have some synchronisation problem
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas da...@debsinc.com wrote:
So this *should* work??
[outgoing]
- exten =
That will work on an outgoing call. Apparently (AFAICS) there is no feature
in Answer to jump to H or continue like the Dial command has.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Tuesday,
Liivo,
I wonder if you are dealing with this general class of issues:
https://issues.asterisk.org/view.php?id=11491
-- Alex
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
I have plugges only 2 lines. That's why the rest is in RED
[r...@pbx ~]# lsdahdi
### Span 1: WCTDM/8 YSTDM8xx REV E Board 9 (MASTER)
1 FXOFXSLS (In use)
2 FXOFXSLS (In use)
3 FXOFXSKS (In use) RED
4 FXOFXSKS (In use) RED
5 FXO
Dan Journo wrote:
- on which linux flavour i should start the installation of asterisk
(CentOs,Fedora,Ubuntu)
right now i am using ubuntu 8.04lts
I use Centos 5.3
CentOS 5.3 works well, though this is somewhat of a religious argument
If you are familiar with Ubadoobie, then why not use
I posted a message the other day
http://lists.digium.com/pipermail/asterisk-users/2009-October/239728.html
about not being able to use newer versions of libpri. I am stuck at
libpri 1.4.1
I posted the pri intense debug span 1 output for a good call and
failed call.
Was wondering what I can do
While this list provides VERY good information, it should not be your only
course of action. You should be looking at the Digium forums, googling
and opening a ticket and/or browsing the issue list. Also check
asterisk-guru and voip-info.org for further information.
-Original Message-
Hi,
I have to set this up for a client, where he could dial multiple extensions
at once, and then put all who picks up into a conference.
I am using a script which does it using originate command. But the originate
commands run one after another, and so it takes a few seconds to call the
Use an AGI that does a Mass originate/call to ring everyone at once. Have
the AGI do an originate loop using a context to dump into the conference and
call it from the dialplan like this:
- exten = s,1,AGI(massconf.agi|ext1|ext2|ext3|ext4|ext5.)
_
From:
Someone? As * is used so extensively with SIP I must've made a _glaring_
mistake in my config (!)
/Rob
Robert Bielik skrev:
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the
Lol
Sent from my iPod
On Oct 27, 2009, at 6:59 AM, Alex Balashov abalas...@evaristesys.com
wrote:
aster...@opensourcesolution.in wrote:
installing asterisk
I am intrigued by your ideas and would like to subscribe to your
quarterly newsletter, as well as attend your biannual leadership
Hi Danny,
This is exactly what I am doing, but it takes a few seconds before all the
extensions are ringing. The loop takes its time.
I need something as quick as Dial(SIP/201SIP/202... which is truly call all
at once, but it connects only two channels, i.e. the first once which picked
up, and
Have you tryed to generate .call files at once ?
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Once the card was configured correctly, have you set on the GUI the correct
port to your zap extension?
On Mon, 26 Oct 2009 05:02:18 -0700 (PDT), PATRICK KANGETHE
patricemb...@yahoo.com wrote:
1. When i connected my analog phone to fxs card, i cannot get dial tone
what could be the problem?
This might be a better application of a call file than an AMI originate.
The AMI originate in this case has to operate in a threaded fashion, whereas
if you created a call file for each extension and dumped them into
/var/spool/asterisk/outgoing, pbx.c would call all of them at once without
the
Hi Alex,
Yes, it's almost the same, except the fact that in my case timestamps
sometimes decrease drastically. In internal network I have Snom 3xx
phones with upgraded firmware, internal leg has no issues, i captured
both legs and phones-asterisk part is ok, the other part,
asterisk-provider
Zeeshan Zakaria escribió:
Hi Danny,
This is exactly what I am doing, but it takes a few seconds before all
the extensions are ringing. The loop takes its time.
Are you originating the calls asynchronously?
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
I think I should try the .call files. I haven't tried them in this
particular scenario yet.
Miguel, what exactly you mean by calling asynchronously? I do 'originate' in
a 'while' loop once I have retrieved all the extensions to dial from the
database.
--
Zeeshan A Zakaria
On Tue, Oct 27, 2009
Zeeshan Zakaria escribió:
I think I should try the .call files. I haven't tried them in this
particular scenario yet.
Miguel, what exactly you mean by calling asynchronously? I do
'originate' in a 'while' loop once I have retrieved all the extensions
to dial from the database.
I mean
Lacking any response I tried to set insecure=invite on both sides. And lo and
behold, the call
gets through.
Now, is this good or bad?
/R
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To
If you aren't doing an explicit async: true, then you are synchronous. Heed
this post as well (Thanks Miguel)
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg231570.html
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Since you are doing peer-to-peer, this should be harmless.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Bielik
Sent: Tuesday, October 27, 2009 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial
Research wrote:
. I saw a nice article on voip-info.org on how to replace voicemail
server for Avaya Definity with asterisk.
Could you send me the link of the article? I'll be looking into doing
this within the next year.
Thanks,
Doug
Hi Doug
See:
Hi all,
Another simple question: does it make sense to use the append option in
MixMonitor (,a) when the codec is gsm? Or it works only when the codec
is an uncompressed one like ulaw, alaw or slin?
Thanks,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
Dear All;
I am facing a problem that all the mobile devices that support SIP and are able
to register with a lot of providers, they are not able to register on my
asterisk. What could be the reason? Any specific thing I have to do?
The used port is UDP 5060
Actually, any SIP Phone can
Try this link
http://www.voip-info.org/wiki/view/Asterisk+Connecting+to+the+Cellular+Netwo
rk
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, October 27, 2009 3:59 PM
To:
I just installed an Asterisknow server
can someone suggest a software to be used for a PC - PC voice comunication
to test in easy way the functionalities of my server.
Thanks in advance for the help
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Xlite softphone??
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of giancarlo
lombardo
Sent: Tuesday, October 27, 2009 4:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Software for
On Tue, 2009-10-27 at 09:34 +, aster...@opensourcesolution.in wrote:
hi ,
i have started reading asterisk book need your guidance.friend as i
am newbie in asterisk so plz plz forgive me if i ask stupid questions.
Installing Asterisk
- on which linux flavour i should start the
On Tue, 27 Oct 2009, giancarlo lombardo wrote:
I just installed an Asterisknow server can someone suggest a software to
be used for a PC - PC voice comunication to test in easy way the
functionalities of my server.
If your PC is running Windows, DIAX is the smallest and easiest soft
phone
Hi there, I have an old Cisco ATA-188-I2-A that I want to revive but with
SCCP (right now it has SIP).
the version i am looking for is ata_03_02_04_sccp_090202_a.zip
i want to do a home experiment with chan_sccp and some recompilations
any links beside cisco to download the firmware?
i do not
On 27 Oct 2009, at 23:29, Erick Perez wrote:
any links beside cisco to download the firmware?
i do not have a valid contract, so cisco does not allow me to
download it.
So you want to pirate it instead?
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Async: True was the solutions to my problem. Thanks for pointing me in
this direction.
--
Zeeshan A Zakaria
On Tue, Oct 27, 2009 at 12:11 PM, Danny Nicholas da...@debsinc.com wrote:
If you aren't doing an explicit async: true, then you are synchronous.
Heed
this post as well (Thanks
If I have a SIP provider (in this case a PBX using SIP trunks), and
I want to send the calling extension number and name as the from in
the SIP invite, how do I set up my sip.conf entry for that provider? I
find the documentation confusing on this point.
callerid=Some Name In From Header 7065551212
Richard Kenner wrote:
If I have a SIP provider (in this case a PBX using SIP trunks), and
I want to send the calling extension number and name as the from in
the SIP invite, how do I set up my sip.conf entry for that provider? I
find the
callerid=Some Name In From Header 7065551212
So the first part is the NAME and the second the number, right?
But my question was how to have that be information from the CALLERID
channel variable rather than a fixed value in sip.conf.
___
--
On 28/10/09 3:52 AM, Danny Nicholas wrote:
This might be a better application of a call file than an AMI originate.
The AMI originate in this case has to operate in a threaded fashion,
whereas if you created a call file for each extension and dumped them
into /var/spool/asterisk/outgoing,
Hi
Now, my Cisco AS5300 sent call to my asterisk, but two problems:
When i call the phone number, i have:
[Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite:
Call from '' to extension '042600' rejected because extension not found.
[Oct 28 06:01:18] NOTICE[12813]:
Ok, so this might seem like a stupid question, but I don't quite
understand how to dial out to the pstn though my T1 from a specific
number. Maybe i'm missing something, but everything I'm reading has
you dial a number from the group but that's not what i'm looking
for. If someone can
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