Hi all,
I have a very simple setup
- 2 ATA modules connected to my Norstar 3x8 extension 25 and 26
- 4 PCI FXO card in my asterisk box, two of then are connected to those
extension 25 and 26.
I have a context for those 2 ata modules simply to get my voicemail.
From my Norstar telephone if i
Worth taking a look as soon as I can migrate my cdr to mysql.. Thx
Guillermo.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Freige
Sent: Miércoles, 23 de Febrero de 2005 03:56 p.m.
To: asterisk-users@lists.digium.com
Subject: RE:
Bob,
To be honest I don't mind who helps, but thanks so far to yourself, Doug
Lytle and Craig Guy. This started off as simply trying to enable CAPI
(Fritz card) on my Mandrake 9.2 pbx which was working perfectly and has
morphed into Mandrake 10.1 with mISDN which is still not working on FXO
let
On Wed, 2005-02-23 at 16:11 -0600, Anton Krall wrote:
I just noticed that quality of .gsm files for using with asterisk is not
that good.. is there any way to make then sound better? asterisks sample
voices sound way better than theones recorded using applications like
wavepad or with asterisk
In article [EMAIL PROTECTED],
Jon Bebeau [EMAIL PROTECTED] wrote:
May I ask which Linux ODBC driver your using on the Asterisk Box to talk to
MS/SQL?
Thanks
I'm not using ODBC. I'm using the cdr_tds module, which speaks natively
to MS/SQL. It's a standard module in asterisk/cdr, recently
For the unknown.
ZAPHFC is a driver that enables the use of a cheap ISDN card to run in TE or
NT mode.
In other words, to run like a standard ISDN terminal to receive and place
calls over a BRI line.
The driver also enables to us a hfc card in NT mode which enables it to
connect to your own
ISDN
Look at ast_exists_extension in pbx.c
I have tinkered a bit around them, but not directly coded to it. In the
app_voicemail.c code I think it is used to make sure the extension is
valid. There might be code in there to see if the registration is
active.
Look further I see that chan_sip.c has a
I did. No joy. Output from Asterisk console below. The parameter is getting
through OK, but same result.
-- Attempting call on Zap/1/650 for application txfax(/tmp/test.fax|caller)
(Retry 1)
Channel Zap/1-1 was answered.
Lauching txfax(/tmp/test.fax|caller) on Zap/1-1
-- Hungup
On Wednesday 23 February 2005 22:21, Razza wrote:
[...]
I'm still concerned that the the /var/log/kernel/errors file states -
Feb 23 17:08:14 asterisk kernel: zaptel: version magic '2.6.8.1-12mdk
686 gcc-3.4' should be '2.6.8.1-12mdk-i586-up-1GB 586 gcc-3.4'
And when I run 'cat /proc/version'
We are seeing a situation where, under busy call volume, Asterisk
hangs up an channel going into a channel bank (via T-1 interface), but
the channel bank does not see the hangup and keeps the channel offhook.
Has anyone seen this problem before? Did you find a solution.
Asterisk(T-1 port) -
here is what I am trying to do:
I currently have 1.0.5 running on our primary box.
Got 40 phones.
HEAD is running on 2ndary box.
Instead of reprogramming 40 phones to use 2ndary box, I want to install HEAD
side-by-side with STABLE.
I need them both on same machine so that I can halt stable, run
Hey Dean,
I followed your instructions and moved forward abit... Before it used to say
SIP 2.0 404 Not Found. But now with your settings I am getting SIP/2.0 407
Proxy Authentication Required.
And yea for some unknown reason the TRUNK name and USER CONTEXT has to be
same don't know why...
But
I tried using sox -v but not any noticable change.
You mentioned I can use wavs for voice prompts (playback and background)
instead of gsm? If this is the case, I do have normalize here so that can be
used.
What I I mean for bad quality is a hissing noise on recordings but
surprisingly enough,
Nitesh do you have the register trunk portion set up as well?
For faktortel is was
xxusernamexx:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 6:05 PM
To: 'Asterisk Users Mailing
On Wed, 23 Feb 2005 13:25:55 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
hehe, maybe I should go ahead and drive myself home and get away from a
computer for the rest of the day. If you are right, I will probably end
Heheh, maybe not just a day!? :)
up totally pissed before the day is
Guys.. Im new to asterisk and this list in particular but if I may (and due
to the fact that some people out there might just be saying: you may not,
I need to add this for you: I dont care), anyways, if I may add a couple
of points here:
1. I agree with the idea that there should be 2 lists,
On Thu, February 24, 2005 10:52, Anton Krall said:
Guys.. Im new to asterisk and this list in particular but if I may (and
due
to the fact that some people out there might just be saying: you may
not,
I need to add this for you: I dont care), anyways, if I may add a
couple
of points here:
Where do these weird ideas come from? :-) Why on earth would it relate
to the dial plan? What difference can you imagine the dial plan might make?
If you want txfax to behave as the calling, rather than the answering,
side you need to specify the caller option. Don't confuse caller with
James,
Are watching the SIP Messaging? SIP Trace on the phone and sip debug... on
the Asterisk box?
The Asterisk should be sending a NOTIFY to the Snom when that hint line is
hit. If you see it in Asterisk, verify that you received it on the SIP
Trace page of the Snom.
When rebooted, the Snom
Duane:
Unfortunately, I didnt get to see your website when I was starting about a
week ago with * but on behalf of future newbies, thank you for taking the
time to actually contribute on help others.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Razza wrote:
I have added 66-zaptel.rules file as you suggested...do I not need
to ref this in a conf file or does udev simply look for any '*.rules'
file in /etc/udev/rules.d/ ?
That would be correct, udev will look for them there.
Which to me whould show some form of mismatch, or is this
Hi All,
I am having trouble getting speex to work on asterisk. I downloaded
1.0.4 from speex.org, download libogg from vorbis. ./configure, make and
make install for both and then recompile my asterisk 1.0.5, yet I am getting
error when trying to load codec_speex.so. Below is the error from
To test the TE410P itself, you can construct a simple cross-over cable
by hacking up a short CAT-5 cable as you describe:
1 - 4
2 - 5
4 - 1
5 - 2
Note that a CAT5 crossover cable will not work. Once you've done this,
set up two spans on your TE410P as you've done it, except that one of
them
On Mon, Feb 21, 2005 at 08:19:58AM +0300, Julius Kidubuka wrote:
Hello,
I am trying to setup an Asterisk GUI with the help of astman(please visit
http://astman.sourceforge.net/am-user-guide.html).
I have installed astman and currently assessing my GUI using;
My configuration is: 1) Channel bank with T1 card. Capi Interface with C4 card,
An IP phone.
I have actually the above problem:
I record a file.
I try to hear it and it gets heared well from some interfaces while from others
is heared with interruptions. Actually I hear well throught the Capi
On Feb 24, 2005, at 1:35, Jonathan Lin wrote:
I can see libspeex.so.1 in /usr/local/lib and it's symbolic linked to
libspeex.so.1.2.0 so the only thing I can think of is the permission.
I
changed the permission to 777 for libspeex.so.1.2.0 just for testing
but
it's still crashing. Has anyone
Jonathan Lin wrote:
Hi All,
I am having trouble getting speex to work on asterisk. I downloaded
1.0.4 from speex.org, download libogg from vorbis. ./configure, make and
make install for both and then recompile my asterisk 1.0.5, yet I am getting
error when trying to load codec_speex.so.
On Tue, Feb 22, 2005 at 01:54:37AM +0100, Philipp von Klitzing wrote:
Hi there,
since I found a couple of reports with complaints concerning zaprtc I
thought that one or the other user might be glad to know that it works
indeed. All that was necessary was to copy all *.h files from /zaptel
My question is how come it was working fine before and after a reboot it
stopped working!? I had the server running for weeks without any problem...
Maybe you updated your Linux, like running yum update, etc
Just a wild guess
___
Asterisk-Users
I've spent the past several weeks reading up and
playing around with Asterisk while I've been waiting for an ISDN card I got on
ebay to arrive so I can really get to business. I'd just like to run my project
ideaa by some of you to hopefully get a little feedback. I aplogize if this ends
up
Hello List,
Can you please point me to the right resources on making multiple sip
phones behind a firewall w/ private address work with asterisk w/c is on
a public network.
I have seen STUN on the grandstream and Xtunnels on X-lite. What is most
deployed by members here with similar setups?
down on a number of unecessary postings, and their assosciated replies from Mr
Critchfield.
Well, that would be bad, because sometimes the answers from Mr
Critchfield just make me roll on the floor laughing.
Keep up the good work Steven ;)
___
Olaf Klein wrote:
Why not just kill yourself, fucking wannabe spammer? DIE DIE DIE
This is *REALLY* offtopic, but Isamar is the founder of Brazilian
AntiSPAM - http://antispam.org.br/ and later http://spambr.org/
Does it matter here? I don't think so, but calling he (or even me) a
spammer is
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hermann Wecke
Sent: Tuesday, February 22, 2005 11:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: FAX
Olaf Klein wrote:
Why not just kill
From my personal experience, the 'weird ideas' come from a lack of
consistent documentation. The caller option, and the use thereof is not
clearly explained anywhere that I can find, and examples of spandsp that are
floating around the WiKi (among other places) erroneously leave this option
I'd be glad to provide this. Where and what do you need?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Wednesday, February 23, 2005 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
* can be used in a CO switch. As long as
you can do DTMF interface.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Waterman
Sent: Wednesday, February 23, 2005
8:46 PM
To:
Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Asterisk
as a voicemail for a
I successfully installed [EMAIL PROTECTED] 0.6, but when i try to load the
wct4xxp module for the TE405P I just got, I get the following:
Found TE410P at base address fb7ffc00, remapped to e0220c00
TE410P version c01a009b
Tried to load 1c8ac800 into , but got 188ac800 instead
Tried to
Shane Burrell wrote:
I'd be glad to provide this. Where and what do you need?
Since I don't know how this is done I don't know *exactly* what is
needed. All I know is it's the GR-1367 stuff. You can read for
yourself and upload anything that might help here:
Hello,
I need you comments on how to configure the operators number 0 after the
digital reception and or if no answer in the incoming call from PSTN.
Your time and effort is very much appreciated.. My colleague also have the
solution for Quintum and Asterisk server interconnection.. maybe we
James,
Are watching the SIP Messaging? SIP Trace on the phone and
sip debug... on the Asterisk box?
The Asterisk should be sending a NOTIFY to the Snom when that
hint line is hit. If you see it in Asterisk, verify that you
received it on the SIP Trace page of the Snom.
When
Hi Kanisha,
After the long time of simulation we are now been
able to send a calls to a Quintum Sip gateway on its PBX ports but not yet on
the PSTN ports. We are glad to share all the config but we still need to make
things work well. BTW, what is your SIP gateway?
Regards,
Jessie
I would like to ask what people think the best way would be to build a
low-power consumption, passively-cooled system.
For example,one could use a fanless-Eden (mini-ITX/EPIA) system, but the
loss of FPU power would limit performance. The obvious choice for FPU
work are the Intels and AMDs, nut
There are so many possibilities here...
First, get hold of every whitepaper that you can find on NAT Traversal in
SIP, so you at least understand the issue.
In my case, with Grandstream phones, I set them to use STUN, and make sure
that they use a dynamic port.
Your ultimate solution will be
Hi Pulu 'Anau,
thank you very much for the advice.. we already did
some part.. and still we are on simulation to prefect it...
thank you very much again..
Regards,
Jessie
- Original Message -
From:
Pulu 'Anau
To: Asterisk Users Mailing List -
Non-Commercial
Hi List,
I just used cvsup to get the latest version of asterisk but I get a
compile error.. :
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\\
What channel bank? I have a Adtran TA750 and the BCU had outdated firmware.
Upgrading the firmware fixed this problem in my case.
Lyle
- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Jim Van Meggelen wrote:
I would like to ask what people think the best way would be to build a
low-power consumption, passively-cooled system.
For example,one could use a fanless-Eden (mini-ITX/EPIA) system, but the
loss of FPU power would limit performance. The obvious choice for FPU
work are the
exten = 691,hint,SIP/691
should do the job. I've got it working with SNOM 190 Phones and actual
CVS-HEAD.
Perhaps there is a problem using the callerid instead of the extension in
the hint?!
Hope, it helps...
-Ursprüngliche Nachricht-
Von: James Bean [mailto:[EMAIL PROTECTED]
On Wed, 2005-02-23 at 21:06 -0500, Time Bandit wrote:
down on a number of unecessary postings, and their assosciated replies from
Mr
Critchfield.
Well, that would be bad, because sometimes the answers from Mr
Critchfield just make me roll on the floor laughing.
Keep up the good work
Perhaps you want to use phpconfig.php Created by p0lar, Dave Packham Rob
Birkinshaw. We use it within our production servers without any problems,
and it does nearly everything, to configure and restart/reload asterisk.
Astman didn't run stable on our servers and after spending some hours
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hecken, Guido
Sent: Thursday, 24 February 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Snom phone hint exten question
exten =
I just upgraded a 7960 from Call Manager to Sip 7.3
Not having a clue, I couldnt load on 7.3 You will need to load on a version 3
sip image, then load on a verison 5 and then goto a version 7. As the doco
says, its multistage
Anyway, the 7960 works beautifully on the asterisk box, have the
On Feb 20, 2005, at 1:26 PM, Brian Roy wrote:
On Sun, 20 Feb 2005 23:16:00 +0900 (JST), Isamar Maia
[EMAIL PROTECTED] wrote:
Ok. I will be burned in fire.. :-)
Or better.. I won't go to the heaven...
You are probably right. But in the the mean time, while you are here
on earth, you will probably
did you try the auto-config? help-aah will tell you
how to do this.
--- BSDR [EMAIL PROTECTED] wrote:
I successfully installed [EMAIL PROTECTED] 0.6, but when
i try to load the
wct4xxp module for the TE405P I just got, I get the
following:
Found TE410P at base address fb7ffc00, remapped
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hecken, Guido
Sent: Thursday, 24 February 2005 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Snom phone hint exten question
exten =
Steven,
YES this is posted as html (when I'm on the road, I can only send html; I agree
that non-html is generally better, but you should notice the disctinct absence
of emoticons and background wallpaper; I suppose I could wait till next month
when I'm home, reply, but run the risk of
Thanks for your contribution Guido!
Do you have a URL I can visit to help me install and configure
phpconfig.php? Otherwise, I'll take a look at phpmyedit and see how ot
works for me.
Rgds,
Julius.
Perhaps you want to use phpconfig.php Created by p0lar, Dave Packham
Rob
Birkinshaw. We use
On Wed, 2005-02-23 at 22:21 +, Razza wrote:
Bob,
To be honest I don't mind who helps, but thanks so far to yourself, Doug
Well, I'll just jump in then :)
I'm still concerned that the the /var/log/kernel/errors file states -
Feb 23 17:08:14 asterisk kernel: zaptel: version magic
Simple cd into your asterisk-src
export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login ; password is anoncvs
cvs checkout phpconfig
Copy all php-scripts and the image folder into your webroot
(/var/www/html/asterisk)
Read/edit the config-files to fit your requirements.
Type in your
Ben Ricketts wrote:
Hi List,
I just used cvsup to get the latest version of asterisk but I get a
compile error.. :
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DASTERISK_VERSION=\\
I want to have a single meetme conference room that interconnects H.323
(Bvideo phone clients and sip/iax audio phone clients.
(B
(BI have already set up for meetme to be shared by sip/iax audio phones and I
(Bhave just now installed open h323 stuff.
(B
(BRegarding to this, I have several
Hello Guys
I have 2 azacall 200 as extensions for a Internet public asterisk box
, I just want to connect point A with point B
I have tried
canreinvite= yes / no
Nat=yes/no
Nat=1/0
commented canreinvite
commented Nat,
I have no audio with other azacall or to other sip device in the same
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