- Original Message -
From: asterisk [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 11, 2005 12:15 AM
Subject: Re: [Asterisk-Users] Re: www.openpbx.org
The other thing that I think many are missing
Hi
I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying
to configure for incoming and outgoing calls from my asterisk
installation. So here is my
On Mon, 2005-10-10 at 14:16 +0800, Craig Guy wrote:
The practical part of the exam showed a distinct USA bias - It was in terms
of T1's and analog zap extensions. I am from Australia, and the exam was in
That is ok, most of this list seems to be the same way regarding the
US/North American
On Friday 07 October 2005 17:48, Michael Stahl wrote:
Roman:
I created two bash scripts called Mail2Fax and Fax2Mail for use with the
asterisk sever.
They leverage the app_txfax and app_rxfax scripts, along with ast_fax.
They make using these apps a lot easier, including being able to mail
On Mon, Oct 10, 2005 at 07:32:31AM +0200, Tzafrir Cohen wrote:
On Sun, Oct 09, 2005 at 07:37:41PM -0400, Steve Gladden wrote:
The debian package installs something else called mpg321 and creates
an alias or symlink called mpg123 to mpg321.
Get the package mpg123 from non-free
That is:
asterisk-users
where can be find zaptel cvs change log ?
thanks
oncemore
[EMAIL PROTECTED]
2005-10-09
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
The original poster's statement about not even receiving any proof that
he was certified is kind of amazing. That's not a certification
by any definition I know of. I would push Digium on that
because they really don't have a leg to stand on if that is true.
If they sold it as a certification then
Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for
quite a while in PTM mode and it was working finely.
Now, I needed more DID/MSN, so I switched to PTP. But now nothing
works anymore :(
I am using Asterisk on Debian Sarge stable and installed Asterisk
Hi, all
I have installed TDM400 card. I can see it is there (lspci).
But Asterisk does not find it.
phonebox2*CLI zap show status
No Zaptel interface found.
I assume that driver is not loaded, but I am sure I have installed it (I
compiled zaptel and then re-build asterisk and did make install
I think you can't use a Fritz Card for PTP. You need an active card. We use the
the beronet ISDN Cards with misdn.
Lionel Riem wrote:
Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for quite
a while in PTM mode and it was working finely.
Now, I needed more
Hello,
Well, now, with the help of mISDN you can, according to http://
www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-
html/x3343.html :
With the introduction of the isdn4linux new mISDN architecture and
it's capi layer, that problem is fixed. chan_capi supports PTP
Hi Tzafrir !
Thanks for your help!! Now it works.
It took some time to find everything and to set up everything, but now it
works.
So I can tell: using asterisk on book pc's with cyrix processors and VIA
chipset compiles fine.
Now I need to check what the performance is like.
thanks
I took the certification in Astricon Madrid, still I have to get any kind of
proof/certificate.
I contacted the testing company and they told me it was just a matter
of time, so probably they are working on this probably those are
just super rapid growing problems.
Regards!
s The
I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS),
no PRIs.
Some users are complaining that they hear clicks and pops on the FXS lines,
generally when they pick up the phone it's noisy. This happens only after a
while, e.g. after a fresh restart of everything,
Hi
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my
Hello list,
I am looking for a way to have multiple remote Windows users download a
package and get connected to *. My idea would be that they run a simple
app, it connects without any setting to an * box (maybe via IAX) and then
people press a button to talk. It would be okay if they had
I don't think you will have any problems at all. I have always used the
lowest setting no matter how long the cable (never had a very long cable
run) and have never run into any problems.
My assumption of the CSU settings are because different CSUs have different
voltage outputs (this is just a
Hi all,
What methods (software or even on paper) would you folks use /
recommend for the purposes of documenting how a dial plan is
constructed? ie. what extensions jump to other extensions, etc? This
is as a means of getting the big picture rather than having pages
and pages of printed
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my
What you are looking for is a pc2phone dialer. This can be preconfigured
with all settings and when it connects to your * it ask for username and
password or just a pin. There are many of these out on the net. Most is
however locked to a provider but you will also find many that you can buy
with
Four years ago, I faced a real dilemma in my business: the Visual Voice PRI
dll had a bug that considered unanswered any call after ringing for 20
seconds. This bug was in fact killing my business, because for international
calling, the setup of the call was already close to 20 seconds on many
What am I doing wrong here? Why is this happening?
libpri is version 1.0.7-1 (debian package)
asterisk is version 1.0.7.dfsg.1-2 (debian package)
zaptel is version 1.0.9.2
-- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack
-- Called g1/0916000739
-- Channel 0/1, span 1
I am trying to setup an h32h channel in Asterisk
Firstly I tried to use chan_h323, but I was not able to compile the
required pwlib ad open323 version under my system (Suse Linux 9.2)
Next I tried to use oh323. I succed in compile and install the
pwlib-Mimas_patch2-src-tar.gz
then
The UTStarcom F1000 with the latest firmware (3.10st) has improved
sound volume over the default firmware shipped with the units.
Also, TFTP configuration works well so you don't have to configure the
units with the keypad. You will need to get the configuration
compiler from your vendor and be
I am running CVS-HEAD-04/12/05-21:44:31 and I am curious if it is
possible to customize the email message sent to the pager email address.
Thanks,
Andy
--
H. Andy Goss
Network Engineer
Network Advocates Inc.
Main: 502.412.1050
DID: 502.992.5933
Mobile: 502.387.8216
[EMAIL PROTECTED]
I am still looking to solve this problem, does anyone have any ideas?
Thanks,
Andy
-Original Message-
From: Andy Goss
Sent: Friday, October 07, 2005 5:37 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] call to a particular 800 number never
Andy,
I may be wrong, but I think that you need to edit the code and
recompile to change the message. I wanted to add a numeric line as
the first line of our system's voicemail pager message so it would
work with numeric pagers as well as text pagers. AFAIK, editing the
code and
hi
how much bandwidth is used for the following codecs
723 r 5.3
723 r 6.3
723 r 8
what i know so far is the
723 r 5.3 uses 5.3 k up and 5.3k down
723 r 6.3 uses 6.3 k up and 6.3k down
729 r 8 uses 8 k up and 8k down
is this correct or is it like the following
723 r 5.3 uses 11 k up and
[EMAIL PROTECTED] wrote:
WARNING[12461]: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol:
_ZNK8PChannel7IsClassEPKc
Oct 10 14:29:20 WARNING[12461]: Loading module chan_oh323.so failed!
Does anybody know which is the problem ?
It seems Asterisk source and binary version do not fit.
What am I doing wrong here? Why is this happening?
libpri is version 1.0.7-1 (debian package) asterisk is version
1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2
-- Executing Dial(SIP/739-5935, Zap/g1/0916000739) in new stack
-- Called g1/0916000739
-- Channel 0/1, span 1
On 10/10/2005, Rich Adamson [EMAIL PROTECTED] wrote:
snip
If you don't have any T1/E1 connections to the outside world, then
pick one channel bank and call it your official source of sync, and
change the above definitions to sync off that channel bank. On all
other channel banks, configure them
i have tellular cell phone plugged to fxo modules ,the
problem that i am facing is when i dial a number on
the fxo modules the call is been answered before it
picked up on the other side , i thought that the
analoge lines do not support the answer switching
feature but not tellular cell phones
If you don't have any T1/E1 connections to the outside world, then
pick one channel bank and call it your official source of sync, and
change the above definitions to sync off that channel bank. On all
other channel banks, configure them to sync off the asterisk card.
snip
Your clicks
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
register = nnn:[EMAIL
On Mon, 2005-10-10 at 13:28 +0200, lenz wrote:
I am looking for a way to have multiple remote Windows users download a
package and get connected to *. My idea would be that they run a simple
app, it connects without any setting to an * box (maybe via IAX) and then
people press a button
On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote:
Dinesh Nair [EMAIL PROTECTED] wrote:
too much divergence and we have two pieces of software competing for each
other.
My guess is that if they succeed, they will diverge significantly.
We will have two pieces of software that
Two asterisk boxes 150 miles apart, both cvs-head as of this morning
(and since Sept 27th), connected via iax2 with low-utilized ds3 internet,
C7960 calls exten on remote system (also C7960), and call goes to VM.
No other calls in either system (eg, no load).
Both boxes have iax
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
register =
Thank you for your respond, please see more detail inline...
Min
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: Friday, October 07, 2005 4:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] TDM02B card
Hi list,
I have a couple of questions related to asterisk billing and the
generation of cdr logs. I've searched the wiki but have not found my
answers, hopefully you guys can help.
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both?
Hi,
Yes, you can use the Fritz! in PTP mode, though only if you are using the
mISDN drivers. The mISDN driver should be called like this:
modprobe avmfritz protocol=34
Craig
- Original Message -
From: Lionel Riem [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Hi all!
I'm experimenting a strange problem in my Asterisk PBX:
I've got an Asterisk pbx in the office: I dial an external number; the dialled
number answers me correctly, but as soon as I press the '*' button (i.e. to
navigate through the menus or to enter a password) my Asterisk box put me on
Hi list (again),
I have another question which I have not been able to resolve from
neither the wiki nor Google.
I've been able to configure a multi-tenant setup of asterisk for 2
small call centers with no problem, by simply playing with contexts
(which I guess is how everyone else is
On Mon, Oct 10, 2005 at 12:28:43PM +0200, [EMAIL PROTECTED] wrote:
Hi Tzafrir !
Thanks for your help!! Now it works.
Now, how would we detect that to avoid needless manual editing of the
CPU?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
On Mon, Oct 10, 2005 at 01:28:17PM +0200, lenz wrote:
Hello list,
I am looking for a way to have multiple remote Windows users download a
package and get connected to *. My idea would be that they run a simple
app, it connects without any setting to an * box (maybe via IAX) and then
I failed to make TDM400 working too myself. But I believed
I passed the the driver stuff... by installing zaptel-1.2.0-beta1.
Inside the package, there is a script zaptel.init that should
take care of loading/unloading the driver.
Min
-Original Message-
From: [EMAIL PROTECTED]
On Sunday 09 October 2005 18:42, Rod Bacon wrote:
Can someone who is knowledgable in the traditional telco space please give
me a layman's explanation (or point me to an appropriate url) of LBO as per
the zaptel configuration file?
Unless something has changed in the last two years, zaptel
Rich Adamson wrote:
snip
One other item to check is to ensure the digium T1 card is on its own
dedicated interrupt. Use 'cat /proc/interrupts' from the system command
line.
It is on one interrupt, first thing I checked when the problem cropped up. One
thing I did notice was interrupt
... which is equivalent to my protocol=0x22 ;)
Nevertheless. I think it was a problem with chan_capi being too old
and not supporting protocol=0x22 layermask=0xf (it would not work
without layermask=0xf).
I am currently trying to get it working with chan_misdn. Will let you
know how it
Ok :)
--
From: Rich Adamson[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Monday, October 10, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] sip register incoming
Mike M wrote:
On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote:
Dinesh Nair [EMAIL PROTECTED] wrote:
too much divergence and we have two pieces of software competing for each
other.
My guess is that if they succeed, they will diverge significantly.
We will
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL. The guys who
developed Mambo have a new product now - Joomla. I am using this and it
appears to be
snip
One other item to check is to ensure the digium T1 card is on its own
dedicated interrupt. Use 'cat /proc/interrupts' from the system command
line.
It is on one interrupt, first thing I checked when the problem cropped up.
One
thing I did notice was interrupt latency when
I'd also like to through the into the discussion my recommendation for
www.xoops.org much better solution than Mambo as far shorter learning
curve.
Also larger development team behind Xoops, particularly as Mambo is now
split messily into two camps and the whole issue about who owns what.
For a
There isn't a way to do it in agents.conf.
That being said though, there are folks that have forgone agents.conf and have used the AddQueueMember and RemoveQueueMember commands via both the dial plan and manager interfaces to work their own agents approach that certainly could be designed to
Rich Adamson wrote:
It is on one interrupt, first thing I checked when the problem cropped up. One
thing I did notice was interrupt latency when doing a 'lspci -v'.. should that
number be 0? If so, does anyone know how to set that at boot time?
I played around a fair amount with the
Hello, i`m carlos, i`m just begining to use Asterisk at Home, so i have learned to configure a several extensions, but now i have a FXO target and i wanna to connect to PSTN, but i dunno how to do. i`d like to receive support from you...thanks
___
Hi Everyone,
Got a setup as follows:
Telco Siemens HiCom 300E Asterisk1 IAX2 Trunk
Asterisk2 Siemens HiPath 4xxx
The solution works except for one problem. Incoming calls from the telco get
redirected to the Asterisk1 box with the correct extention, only if there is
a
On Monday October 10 2005 08:18, Kanishka Somaratne spake:
hi
how much bandwidth is used for the following codecs
http://www.voip-info.org/wiki/index.php?page=Bandwidth+consumption
--
Joey Kelly
Minister of the Gospel | Linux Consultant
http://joeykelly.net
I may have invented it, but
I've been asked to forward this announcement to the list. It's a little
short notice as the meeting is this Wednesday night. I'm one of the
presenters as well :-)
From: Gary Poster [EMAIL PROTECTED]
Date: October 10, 2005 11:51:10 AM EDT
To: zope-announce@zope.org,
I'm having slight problems with outgoing audio quality on Zap channels.
People hear an interrupted voice.
Can anyone help..?
Regards,
Fabrizio Mazzoni
Macron SPA
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Hi all.
Just as a quote note, can I thank everyone on this list. I find my
self finding pretty much every answer I am looking for on here. And a
big thanks to all thoughs helping us out. Mass Respect :)
Ok, I'm using a SIP provider (SipGate UK) to do my international
dialing etc, working great
oner asterisk wrote:
Hi all,
I would like to add indication tones ,
What I did is
enter data to zonedata.c and indications.conf
than compile zaptel. and restart asterisk.
But it's not working what else I should do ?
Regards,
Öner
did you check that the new tones are loaded in
BJ,Thanks for the prompt response. Both my clients work by using the AgentCallBackLogin so that * can send queued calls to them regardless of which SIP phone they're sitting on (sorry I didn't include this in my original email)You mean to say that if I use AddQueueMember, I could do the same and
Thank you for your reply and your help.
I am still confused here and apologize.
To some degree I still do not know what I am doing.
We use 2 ITSP's and one of them we have multiple SIP accounts
on so I will not be able to do this by IP address.
For incoming calls we use a register line in the
On Mon, 10 Oct 2005 11:01:12 -0300, Ivan Stepaniuk [EMAIL PROTECTED]
wrote:
On Mon, 2005-10-10 at 13:28 +0200, lenz wrote:
I am looking for a way to have multiple remote Windows users download a
package and get connected to *. My idea would be that they run a simple
app, it connects
Hi all
I have configured an ISDN bri card to work in TE mode using chan_mISDN in
asterisk. I can both place and receive calls through my ISDN line with no
problems. I am trying to restrict sending my caller id (CLIR) but I don't
seem to find how to do it. Does anyone know how to restrict
For some reason, every morning at 8:30 I get a call on my main
extension. When this call is picked up, it promptly
disconnects. Is there some sort of Wake up call or something
that may inadvertently be set in * that could be causing this? It
has been happening for quite some time, and I always
Steve Gladden wrote:
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
I am running CVS Head i686 running Linux on 2005-06-30 22:55:14. I have
SIP Buddies installed using MySQL.
If I try to set up a ATA that has 2 two phone lines (resulting in 2
lines on 1 IP address), my second line can never authenticate to dial out.
I ran ethereal and found that Asterisk is
I have 3 different SIP extensions in my DIAL string.
i.e. I have HOME_PHONES_TO_RING=SIP/2000SIP/2001SIP/2002
so in my Extensions file i have Dial(${HOME_PHONES_TO_RING},30,tTr)
So... when the home phone line rings, all three phones ring.
Anyways.. the problem is.. in the CDR log, sometimes
Hi all
I try to get the caller id of a incoming call through a X100P generic card.
I have tried many configuration on the zapata.conf, but i never succeed
to have a correct CALLERIDNUM.
What is the cid signaling provided by FranceTelecom (v23 ?)
Is there some specific stuff to do ?
Could you
For some reason, every morning at 8:30 I get a call on my main extension.
When this call is picked up, it promptly disconnects. Is there some sort of
Wake up call or something that may inadvertently be set in * that could be
causing this? It has been happening for quite some time, and I
Hi guys, im using realtime and I want to show registered users or
online users on a webpage and offline users. Im taking regseconds
field to make this happend
If regseconds value is 0 then user appers offline, it regseconds is
something else then its online, but sometimes this works and
Miguel Cavazos wrote:
Hi guys, im using realtime and I want to show registered users or
online users on a webpage and offline users. Im taking regseconds
field to make this happend
If regseconds value is 0 then user appers offline, it regseconds is
something else then its online, but
I've already sunk several hours into this without any
real progress, so I'd really appreciate any help My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go out to a PSTN
Setup the two ports completely separately.
Each should have it's own entry in realtime with a unique username.
-bill
On 10-Oct-05, at 1:15 PM, Dave Wise wrote:
I am running CVS Head i686 running Linux on 2005-06-30 22:55:14. I
have SIP Buddies installed using MySQL.
If I try to set up a
I don't know if this will help you, but we had the same problem, we also
have Polycom 500s and I changed the pickupexten to *9 (anything other
than *8), because I read somewhere that for some reason Asterisk has a
problem with this feature and *8. It worked for us.
Alberto
-Original
Hello list,
today I have been busy playing with addQueueMember, and it is well known
that it does not log to the queue_log file.
The answer - bad as it may seem - is to add a fake queue_log data for each
logon and logoff. This was covered previously by
In data Mon, 10 Oct 2005 18:57:20 +0200, Michael Van Donselaar
[EMAIL PROTECTED] ha scritto:
Rather than recompile with presets, you'd probably want to change the
reg keys
used in the installer. When I was first developing iaxComm for family
and
friends, I distributed the executable with
When I added
group=1
callgroup=1
pickupgroup=1
under each extension then it worked. I assume it is the pickupgroup=1 that
did it. I will experiment to see.
Angus
- Original Message -
From: Alberto Risco [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Has anyone successfully managed to get outbound CallerID to
work on outbound calls through VOXEE?
On CallerID I mean NUMBER ( I know how the PSTN works)
My callees keeps getting private call,
of course the call is blocked when callee has anonymous call block.
I have searched through the
In data Mon, 10 Oct 2005 14:03:55 +0200, tim panton
[EMAIL PROTECTED] ha scritto:
Yep, I'm working on such a thing.
I have a demo version running at http://www.westhawk.co.uk/software/
faceless/CallUs.html
You don't even need to install it, it runs in the user's browser.
( you will need IE6
in sip.conf
[yourextensioncontext]
callerid=1234567890 1234567890
in extensions.conf
[voxee]
exten = _1NX,1,SetCallerID(${CALLERID} |a)
HTH
Best Regards
Greg Cirino
Spam and Virus Free Email
included with every email account
Cirelle Enterprises Inc.
25 Indian Rock Rd #421
Windham
Wolfgang wrote: -
I've already sunk several hours into this without any
real progress, so I'd really appreciate any help My
task is simple -- establish a connection between a
softphone on XP ProSP2 to a Asterisk server on Linux
FC4 over a LAN through a Netgear router. The server
will then go
Im getting this warning on the CLI.
Please im having problems getting extensions to register while using the realm instead of the IP address.
Oct 10 20:17:15 WARNING[3105]: chan_sip.c:11178 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 0
Can anyone
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it shouldn't
matter... unless there's something else going on that
I don't know about.
Thanks
David:
Also port 1:2 is a good idea to forward to the server as well..
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well -- however, as the
client isn't going out past the LAN, it
Hello List,
I have a problem with my Oh323 install on 1.2Beta.
I have 1.2Beta installed on CENTOS 3.5 and downloaded the following
packages:
pwlib-Mimas_patch2-src-tar.gz
openh323-Mimas_patch2-src-tar.gz
asterisk-oh323-0.7.3.tar.gz
However when I go into the pwlib_Mimas_patch2 and
run a
Hi list!
Do anybody has success histories about using the Beronet app_saynumber
application? For those that are hearing about for the first time, it´s a
piece of software that enables the use of another language in say_number
commands in asterisk dialplan or AGI scripts.
Link to download:
Dan Journo wrote:
Can anyone shed some light on this?
In your sip.conf
auth=md5 was the cause for me.
Doug
--
Ben Franklin quote:
Those who give up essential liberties for temporary safety deserve neither liberty
nor safety.
___
David:
Also port 1:2 is a good idea to forward to the server as well..
Only needed for SIP. 4569 is all that is required for IAX2.
David,
Yes, I've also forwarded port 4569 to the server.
Since the router is forwarding to the server, I cannot
forward it to the client as well --
Are you calling from a soft- or hardphone on a network with a high
amount of latency? If your (for example) SIP phone can't deliver voice
packets to asterisk in time for asterisk to put them where they belong
in the Zap channel, things like this might happen. Usually the
interruptions could
On Mon, Oct 10, 2005 at 04:44:16PM -0300, Ricardo Poppi wrote:
Hi list!
Do anybody has success histories about using the Beronet app_saynumber
application? For those that are hearing about for the first time, it´s a
piece of software that enables the use of another language in say_number
On Mon, Oct 10, 2005 at 11:10:59AM -0400, Kanuri, Seshu (Company IT) wrote:
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL.
It depends who you ask.
I thought to add that back in after i removed it because i was having other problems
Any idea why, when i use the IP address for the realm/domain on the SipPhone, it connects ok. But when i use the domain name, it doesnt authenticate?
Thanks
Dan
On 10/10/05, Doug Lytle [EMAIL PROTECTED] wrote:
On Mon, Oct 10, 2005 at 10:27:19AM -0400, Min Qiu wrote:
Thank you for your respond, please see more detail inline...
Min
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: Friday, October 07, 2005 4:57 PM
To:
On Mon, Oct 10, 2005 at 07:25:59PM +1000, Rudolf Ladyzhenskii wrote:
Hi, all
I have installed TDM400 card. I can see it is there (lspci).
But Asterisk does not find it.
phonebox2*CLI zap show status
No Zaptel interface found.
I assume that driver is not loaded, but I am sure I have
Hi!
I have problem with my AAH. I have set up a sip
channel. It works perfect both ways with one exception. When someone calls in I
only get 1 signal. The caller have normal ringtone until message is played.
Anyone who can help?
Regards
Anders Svensson
Kanuri, Seshu (Company IT) wrote:
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL. The guys who
developed Mambo have a new product now - Joomla. I am
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