Hi,
I'm waiting for some hardware phones, so I missing some experience in such
cases. But what you can do is to launch sip debug in the Asterisk's
console, and watch at the logs. You might have a codec issue.
Regards, Jean-Christophe
- Original Message -
From: [EMAIL PROTECTED]
To:
Title: Message
Has anyone
ever heard of such a beast? do they exist? (soft or hard
phone)
I am
referring to the encrypting of the RTP data as the SIP headers will need to be
read by asterisk still
Bryan
Nolen
Lead Developer
http://Arc.Net.AU
http://cdonline.com.au
Hi.
Do you have mpg123 installed and mpg123 in /usr/bin ? I have moh
working with snom200 and there was no issue to have them working.
I even put an extension in my extensions.conf so the user can dial
it an hear the music, cause the snom200 has call waiting they don't miss
calls
On 22/10/2003 1:42, Chris Albertson wrote:
IMAP would work as would an NFS mounted maildir. But I still
would prefer a DBMS based store to support some voicemail features
such as.
[...]
For all of the above a simple SQL query would do the trick
The user interface could be either menues
So, Tilghman has put a particularly useful patch in the bugtracker:
streaming music-on-hold is now supported. You can now specify .mp3
streams to be played back as MOH in the various places where MOH is
used. Hopefully, Mark will install into the main CVS tree shortly.
Hello,
i have asterisk with chan_capi ( AVM Fritz card ) working, isdn line
BRI, setup to point-to-multipoint.
working fine, but when both channel on interfaces are used and somebody
wants to call in - the response is ha-la-li - sounds like msn or
called number does not exists. i think that right
Hi Bryan,
I am aware that the IETF have an Internet Draft in the pipelines for SRTP which can
provide encryption and there is a lib out there available at:
http://srtp.sourceforge.net/srtp.html
I guess the real question would be if there is any intension to include this (or an
equivelant) in
John Brown (CV) wrote:
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time
to develop, test and
I'm not sure if it would be really practical to have a built in switch (although
useful) within the phones. You really don't want your phone worrying too much about
switching other ethernet frames whilst a call is in progress, you will probably then
run in to queueing problems as you need to
My perspective is Yes, there is a LOT of interest! but there are no
desktop phones that are capable of using SRTP at this point. The
IETF is being painfully slow about making this an RFC, and the drafts
have unintentionally expired. sigh
If you feel like prodding your favorite VoIP phone
Hi there
I've installed a 12 port MGCP gateway, (Hitron MDU-5612), which works ok most of the
time. Sometimes when talking to the outside world, (via a TE410P), the line gets
disconnected. I think its related to MGCP because I've also setup some SIP devices
which doesn't behave like this.
Chee Foong wrote:
Hello,
I have been using IAX to serve clients endpoints for a while with no
problem.
But recently, to increase the bandwidth to the Asterisk server, I add
another network interface card to my Asterisk server which connected to a
different service provider that I currently have.
Hello,
I have been using IAX to serve clients endpoints for a while with no
problem.
But recently, to increase the bandwidth to the Asterisk server, I add
another network interface card to my Asterisk server which connected to a
different service provider that I currently have. Both of my nic is
IMAP would work as would an NFS mounted maildir. But I still
would prefer a DBMS based store to support some voicemail features
such as.
[...]
For all of the above a simple SQL query would do the trick
The user interface could be either menues built in extensin.conf or
web based. or
Once you go this route, you can ignore the local filesystem problem and
But you still want handset playback, so you'd want the ability to get the
next message from the mailbox. If you still store a local copy you
haven't solved the defragment problem.
Why store a local copy?
Speed,
On Wed, 2003-10-22 at 02:31, John Todd wrote:
So, Tilghman has put a particularly useful patch in the bugtracker:
streaming music-on-hold is now supported. You can now specify .mp3
streams to be played back as MOH in the various places where MOH is
used. Hopefully, Mark will install into
Title: RE: [Asterisk-Users] A software FAX modem
-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, October 21, 2003 8:23 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] A software FAX modem
Hi,
Interesting. I am using RH9 for
So I've been on the Asterisk list for a long time. Mostly looking to set
it up as a cheap alternative to a commercial MeetMe bridge, as a for-fun
project.
I keep noticing the Wildcard FXO X100P sorta seems like a Winmodem.
Winmodems are basically a sound card chip tied to a pots port, where the
On Wed, 2003-10-22 at 05:06, Adam Williams wrote:
Once you go this route, you can ignore the local filesystem problem and
But you still want handset playback, so you'd want the ability to get the
next message from the mailbox. If you still store a local copy you
haven't solved the
Hi all,
can somebody explain me how exactly the type, host, permit and deny
option in sip.conf play together?
Where is the difference between user and peer ?
I want configure SIP so that it is only from specified net section possible
to make a call.
I have tried following:
[test]
type=peer
Anybody know if a phpconfig readme install guide is available.
Regards
Adrian Brown
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.529 / Virus Database: 324 - Release Date: 16/10/2003
Hi
I am using two gnophones with local asterisk server.The call is getting
through.But in one of the gnophones,no sound is coming.when i boot this PC
in Windows,then sound card is working perfect.But when in Linux,when i run
the gnophone application,no sound is heard.What could be the possible
Adrian Brown wrote:
Anybody know if a phpconfig readme install guide is available.
Regards
Adrian Brown
Not that I know of, what problems are you having setting it up?
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Hi,
I'm trying to install Asterisk onto a fresh install of Slackware 9.1...
I've installed all packages in A, AP, D, E, F, K, L, N... So basically
what's needed for a text based system with development, networking,
docs, libraries.. No X-Windows, no games, no TCL/TEX etc.
Following the commands
Has anyone attempted to translate short text messages into to audio
file (or something like that), and send the result to a predetermined
pbx/cell number via asterisk?
I'm thinking in terms of an external application (something like an app
that watches syslog messages) initiating a system
/ld: cannot find -lXext is your actual error,
Looks like something to do with X windows?
Or go into the makefile and tell it not to make pbx_gtkconsole, i dont think
its essential.
Panny
- Original Message -
From: Christopher Lee [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
=CUT=
In file included from /usr/include/gtk-1.2/gtk/gtkarg.h:31,
from /usr/include/gtk-1.2/gtk/gtkobject.h:31,
from /usr/include/gtk-1.2/gtk/gtkaccelgroup.h:35,
from /usr/include/gtk-1.2/gtk/gtk.h:32,
from
I keep noticing the Wildcard FXO X100P sorta seems like a Winmodem.
Winmodems are basically a sound card chip tied to a pots port, where the
Winders driver takes on the MO/DEM functions in software. I've always
despised them as a data communications device but the other uses of the
cards do
Thanks very for the rapid responses Andrew and Panny!
I went and installed the base packages from the X set for x-windows, and
recompiled asterisk successfully, so that's all good, but you did get me
thinking of removing gtkconsole from the Makefile. However since it's
compiled I won't try and
No particular problem just thought it would see if an install guide was
available before I had a go at installing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Wednesday, 22 October 2003 9:17 PM
To: [EMAIL PROTECTED]
Subject: Re:
We tried to use it witch our AVM Fritz!Card with chan_capi but
asterisk always crashes after our fax-machines shows the ID of the
soft-fax (12345678). Here's a backtrace:
I just found out, that it was my own fault. The apps were linked
against an older version of libtiff (3.5.5) which was
On Wed, 2003-10-22 at 06:04, [EMAIL PROTECTED] wrote:
Hi
I am using two gnophones with local asterisk server.The call is getting
through.But in one of the gnophones,no sound is coming.when i boot this PC
in Windows,then sound card is working perfect.But when in Linux,when i run
the gnophone
WipeOut wrote:
John Brown (CV) wrote:
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new features take time to
On Wed, 2003-10-22 at 07:24, Rich Adamson wrote:
Has anyone attempted to translate short text messages into to audio
file (or something like that), and send the result to a predetermined
pbx/cell number via asterisk?
I'm thinking in terms of an external application (something like an app
Subject: Re: [Asterisk-Users] Call Waiting on SIP phones
Paul, I applied the patch successfully last night on the CVS from last night. I set
the incominglimit=1 in sip.conf. I am still getting the ring in the Grandstream
phones.
I posted a bug report on the bugtracker. If you would like
On Wed, 22 Oct 2003, Steven Critchfield wrote:
Judging by your behavior with this post, I'm betting you are either a RH
or Mandrake user, and may have ALSA running with ARTSD also. So you may
be in for a little bit of work.
Somebody got out of bed on the wrong side this morning, or was this a
Comments inline
Thomas Haeger wrote:
Hi all,
can somebody explain me how exactly the type, host, permit and deny
option in sip.conf play together?
Where is the difference between user and peer ?
I want configure SIP so that it is only from specified net section possible
to make a call.
I have
Yes, a winmodem has all the hardware you need to do the job - however that
is not the entire picture.
Unless the card happens to contain a dsp, or interface chip whose
specifications are public you might as well give up.
This is a similar problem to having an ethernet card and a driver, but
Stephen R. Besch wrote:
WipeOut wrote:
John Brown (CV) wrote:
Hi List,
I had a wonderful meeting with GS's President last week
and he is very interested in feedback on what top features,
functions, bugs the community would like to see in upcoming
firmware.
Please keep in mind that adding new
On Wed, 2003-10-22 at 08:21, Michael T Farnworth wrote:
On Wed, 22 Oct 2003, Steven Critchfield wrote:
Judging by your behavior with this post, I'm betting you are either a RH
or Mandrake user, and may have ALSA running with ARTSD also. So you may
be in for a little bit of work.
Can you _please_ trim the quoted text? There's absolutely no reason to
quote the entire post you're replying to, signature lines and all... +2
points for bottom-posting though. :-)
I still think HTTP is a better option.. There is far more control
available in terms of securing it
Dnia wto 21. padziernika 2003 22:43, Michael Bielicki napisa:
On Tuesday 21 October 2003 9:45 pm, Witold Krecicki wrote:
1st. - I'm from Poland, we don't have (yet, and hopefully forever)
software patents.
Wrong. Not only wrong but copying software illegally and using it
especially in a
http == hyper text transport protocol
tftp == trivial FILE trasfer protocol
thus using tftp to do updates seems better. Its also
a smaller foot print code wise and in boot loader thats
important.
tftp servers are available,
On Wed, Oct 22, 2003 at 08:58:33AM +0100, WipeOut wrote:
John Brown
On 22/10/03 08:37, WipeOut wrote:
You should build you Asterisk box as a dedicated system not as a desktop
with and Asterisk server built in.. :)
This is certainly true for production use. However, I'm also happily
running Asterisk for development and testing on my desktop machine under
On Wed, Oct 22, 2003 at 02:24:57PM +0100, WipeOut wrote:
Here is another thought that I haven't heard mentioned...
How about changing the TFTP upgrade in favour of HTTP upgrades and
config file retrieval.. I am sure almost everyone has an HTTP server
available to them but I doubt many
John Brown (CV) wrote:
http == hyper text transport protocol
So are the entries on your hard drive with a .htm or .html extension not
files? (sorry a little sarcastic I know)
tftp == trivial FILE trasfer protocol
thus using tftp to do updates seems better. Its also
a smaller foot print code
Hi Steve,
Think I understand the * side of that well enough to know it can be done,
but not sure about the text-to-voice app. Anyone have any ideas?
Yes. As part of our need to be proactive about problems instead of
reactive, my company has built a notification system around our apps
John Brown (CV) wrote:
right, adding HTTPS and HTTP to the boot loader would cause that
to inflate and possibly be to big to deal with.
True..
so enable tftp and put a couple of ipfw statements on the box
to limit who can tftp from/to you.
Could be made to work but most IP's are dynamic..
Andrew Kohlsmith wrote:
Text trimmed.. :)
Who in their right mind is putting these phones on the open Internet
An example.. sipphone.com
, and if
they're not, then why is TFTP that big a problem? TFTP's actually quite a
standard option in most networking equipment for pulling down new
On Wed, 22 Oct 2003, John Brown (CV) wrote:
http == hyper text transport protocol
tftp == trivial FILE trasfer protocol
Based on this definition we could suggest that the web should only consist
of a few html files as a jpeg clearly isn't hypertext.
I suspect the reason HTTP was proposed
On Wed, Oct 22, 2003 at 03:15:27PM +0100, WipeOut wrote:
http is a bad idea imho. I don't want to have to carry around
a web server on my laptop, or have to have my customers config
a web server to deal with updating their phone.
I would think setting up a web server would be easier than
On Wed, 2003-10-22 at 10:07, Rich Adamson wrote:
Hi Steve,
Think I understand the * side of that well enough to know it can be done,
but not sure about the text-to-voice app. Anyone have any ideas?
Yes. As part of our need to be proactive about problems instead of
reactive, my
Hi
I am interest with asterisk, but I like download It and the link is break
Please, Any people give me this software
tanks for all
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Hi everyone,
I need to build a machine capable of running at least 30 G.729 channels
with lots of room to spare because it will be doing some other CPU
intensive tasks also.
I've seen Mark's post about being able to run 60 channels on a dual 1.8
Xeon, but that unfortunately raises more
On Wed, 22 Oct 2003, Michael T Farnworth wrote:
I suspect tftp probably has a simple protocol too. Maybe support could be
added for http as well as tftp.
I take this back, as a protocol tftp is hideously complex compared to
http and would take a lot more code.
However tftp is based on tcp
I am looking for a SIP carrier to handle wholesale residential traffic.
Standard LEC services in the US. Anyone have any suggestions?
Thanks,
Stephen
Stephen Dolloff
DLS Internet Services
847-854-4799 x256
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ok, ive just got a wireless internet connection installed at my home. its
pretty much the only way to get a stable connection here in belgrade - so
im happy with it. now the next trick is to get asterisk working bearably
with it. unfortuantely its using a modified version of the 802.11b spec
Figured the group would like to hear this. I just faxed a sample
document from a real fax machine to asterisk semi successfully. I'll
consider it just semi successfully for now because either I haven't
found a viewer that puts the image in proper aspect ratio or the storage
is screwy. I'm thinking
Resending this. Any help appreciated.
Thanks,
-B
- Original Message -
From:
Balaji NJL
To: [EMAIL PROTECTED]
Sent: Sunday, October 19, 2003 6:36
PM
Subject: [Asterisk-Users] newb - want to
create a Dialpad like system
Hi all,
i am planning to
On Wed, 2003-10-22 at 09:40, Chris Ziomkowski wrote:
Hi everyone,
I need to build a machine capable of running at least 30 G.729 channels
with lots of room to spare because it will be doing some other CPU
intensive tasks also.
Maybe you need to become more aquainted with asterisk. Pushing
Lic. Edwin Mamani Z wrote:
Hi
I am interest with asterisk, but I like download It and the link is break
Please, Any people give me this software
tanks for all
Best bet is to use the CVS version to get started..
Follow the instructions on..
http://www.asterisk.org/index.php?menu=download
Dnia pi 18. lipca 2003 10:21, Jeremy McNamara napisa:
The actual error occurs before this spot in the trace. The most likely
cause is codec negocation failure.
I've got the same problem
My log is at http://www.culm.net/oh323 (42kb), if someone could take a look at
this I would be really
Ouch.. you hit one of my pet peeves.. See below.. This is not meant to be
a FLAME but rather SOAPBOX.
Robert
John Brown (CV) wrote:
http == hyper text transport protocol
So are the entries on your hard drive with a .htm or .html extension not
files? (sorry a little sarcastic I know)
***
However tftp is based on tcp rather than udp so it requires less complex
networking support.
Most tftp implementations use udp (not tcp) with an added layer to
identify missed or out of order packets. Slightly more complex then
one would think, but not all that bad.
On Wed, 2003-10-22 at 15:38, Michael T Farnworth wrote:
On Wed, 22 Oct 2003, Michael T Farnworth wrote:
I take this back, as a protocol tftp is hideously complex compared to
http and would take a lot more code.
RFC 1350 (tftp v2): 11 pages
RFC 2616 (http/1.1) : 114 pages
josh.
--
Has anyone tried installing * on a box with two eth interfaces which is
acting as a NAT box? I have only one IP at this point and I would like
to get * working without all of the NAT issues. My idea is to run * on
my gateway (which is also running the firewall and masquerade services).
All of my
hello,
I have jsu configured my first Asterisk PBX and it works well.
In our company we have alose one Cisco AS5300.
How can I mmake Asterisk to forward calls, which have first digit 0 to
that Cisco AS5300.
Our gateway is allready configured to handla that calls.
best regards
Check out the software at
http://www.xten.com/.
Their XTen-Web and XTen.NET products may help you out. Allowing people to
dial a landline is actually quite simple, and can definately be done with
Asterisk.
--Ernest
At 08:00 AM 10/22/2003, you wrote:
Resending
this. Any help appreciated.
On 22-10 15:38, Michael T Farnworth wrote:
In fact I believe a SIP client doesn't have to support TCP, but
fortunately I believe the Grandstream does.
RFC3261 compliant SIP clients must support TCP.
Jan.
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On Wed, 22 Oct 2003, Michael T Farnworth wrote:
However tftp is based on tcp rather than udp so it requires less complex
networking support.
Replying to own email here, which is bad I am told, but I did make a
mistake, I meant to say tftp is based on udp rather than tcp.
Michael
Adam, a patch would be fantastic. I am using chan_h323. I'm under the
impression that the problem had to do with my system being scsi based
instead of IDE based - the license installer read something off of the IDe
bus.
Message: 6
From: Adam Hart [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
On 22 Oct 2003, Josh Howlett wrote:
On Wed, 2003-10-22 at 15:38, Michael T Farnworth wrote:
On Wed, 22 Oct 2003, Michael T Farnworth wrote:
I take this back, as a protocol tftp is hideously complex compared to
http and would take a lot more code.
RFC 1350 (tftp v2): 11 pages
RFC 2616
TFTP config would be fantastic, but right now my #1 piss-off is that I have
to dial a phone number twice or sometimes three times for the phone to take
all the digits. Someone on the list said it was like a manual typewriter
and I agree 100%
Regards,
Christopher J. Wolff, VP, CIO
Broadband
I think that is about the only thing that _will_ work if you
want Asterisk to make and recieve SIP calls to the public
Internet.
If your system were larger you could justify _two_ asterisk servers.
One on the fire wall as you've proposed and the second inside
in back of NAT. All SIP calls
Hi Steve,
This is definately the way to go when you are dealing with NAT.
I have simulair setups running like this myself and it works perfectly.
Only think you need to do in the sip.conf entries is add
canreinvite=no
This will force any sip-calls from the outside to be routed thru *.
--- Steven Critchfield [EMAIL PROTECTED] wrote:
On Wed, 2003-10-22 at 09:40, Chris Ziomkowski wrote:
Hi everyone,
I need to build a machine capable of running at least 30 G.729
channels
with lots of room to spare because it will be doing some other CPU
intensive tasks also.
I hate
Christopher J. Wolff wrote:
TFTP config would be fantastic, but right now my #1 piss-off is that I have
to dial a phone number twice or sometimes three times for the phone to take
all the digits. Someone on the list said it was like a manual typewriter
and I agree 100%
This is my biggest
No it's not broken, I just did a CVS update recently.
Try downloading from CVS and then and if you have
problems quote the error messages in your e-mail.
--- Lic. Edwin Mamani Z [EMAIL PROTECTED] wrote:
Hi
I am interest with asterisk, but I like download It and the link is
break
Hi Chris,
What on earth are you refering to?
Regards,
Steve
Chris Albertson wrote:
This whole argument is moot because there IS a free g.729
implementation. Actually it is a zero cost license to the
source code. Exactly what was asked for.
--- Steve Underwood [EMAIL PROTECTED] wrote:
On Wednesday 22 October 2003 07:49, Steven Critchfield wrote:
Judging by your behavior with this post, I'm betting you are either
a RH or Mandrake user, and may have ALSA running with ARTSD also.
So you may be in for a little bit of work.
If you're running artsd, the solution is very simple.
Hi Steven,
I think I have the tagging right for the aspect ratio. A lot of display
software gets it wrong, including some well regarded things like the
GIMP. KFax displayed my fine and standard test FAXes properly.
Steven Critchfield wrote:
Figured the group would like to hear this. I just
Steve Sobol wrote:
Christopher J. Wolff wrote:
TFTP config would be fantastic, but right now my #1 piss-off is that
I have
to dial a phone number twice or sometimes three times for the phone
to take
all the digits. Someone on the list said it was like a manual
typewriter
and I agree 100%
Is app_meetme broken?
I seem to get invalid conference number all the time :(
Panny
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Hi Tjardick,
Do you mean that * will be used as a proxy by the way ? For the tests I have
made, Asterisk tries to put both phones in relation together. Did I
understand right ?
Jean-Christophe
- Original Message -
From: Tjardick van der Kraan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
It just struck me that the easiest improvement would be to drop the name
BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests
poor quality. The phones need a name which implies 'High Quality'.
The current name resulted in my wife bursting into laughter and saying 'I
can't
At 09:19 AM 10/22/2003 -0700, you wrote:
I hate to say it, but jumpping off into a 100 channel PBX is not
the way to go with Asterisk. Build a 1x1 PBX first on an old
Pentium 500. get this to work then try adding SIP phones then
add some other features. After you've spent some time you will
not
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Panny
Malialis
Sent: Wednesday, October 22, 2003 11:43 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Meetme
Is app_meetme broken?
I seem to get invalid conference number all the time :(
Panny
I am trying to music on hold but I am having all sorts of problems with it.
I am running RH9 and the latest version of Asterisk as of yesterday.
Here is what I did to test it:
1. I manually deleted the mpg123 softlink to mpg321.
2. I downloaded mpg123-0.59r-1.n0i.src.rpm, compiled and installed
Panny Malialis wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
You have to have some Zaptel device installed. Like wcfxo or ztdummy
Jeremy McNamara
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On Wed, 2003-10-22 at 18:50, Michael T Farnworth wrote:
It just struck me that the easiest improvement would be to drop the name
BudgeTone, because 'Budget' tends to imply cheap, which in turn suggests
poor quality. The phones need a name which implies 'High Quality'.
Hadn't thought of that
On Wed, 22 Oct 2003 17:43:08 +0100, Panny Malialis [EMAIL PROTECTED]
wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
(Is there a FAQ for this yet?)
1) Did it work before? If so, what changed?
2) Do you have zaptel hardware installed and working, or the
On Wed, 2003-10-22 at 11:32, Steve Underwood wrote:
Hi Steven,
I think I have the tagging right for the aspect ratio. A lot of display
software gets it wrong, including some well regarded things like the
GIMP. KFax displayed my fine and standard test FAXes properly.
kfax works well as you
Jeremy McNamara wrote:
Panny Malialis wrote:
Is app_meetme broken?
I seem to get invalid conference number all the time :(
You have to have some Zaptel device installed. Like wcfxo or ztdummy
See
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
for more information!
/O
--- Chris Ziomkowski [EMAIL PROTECTED] wrote:
At 09:19 AM 10/22/2003 -0700, you wrote:
I hate to say it, but jumpping off into a 100 channel PBX is not
the way to go with Asterisk. Build a 1x1 PBX first on an old
Pentium 500. get this to work then try adding SIP phones then
add some other
Yup, my problem was that I switched from ztdummy to zaprtc
I rebooted the box and forgot to load zaprtc, doh!
Thanks for the help :)
Panny
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 6:47 PM
Subject: Re:
Can you _please_ trim the quoted text? There's absolutely no reason to
quote the entire post you're replying to, signature lines and all... +2
points for bottom-posting though. :-)
I still think HTTP is a better option.. There is far more control
available in terms of securing it especially
Do you have ztdummy or zaptel device in your system?
- Original Message -
From: Panny Malialis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 22, 2003 1:43 PM
Subject: [Asterisk-Users] Meetme
Is app_meetme broken?
I seem to get invalid conference number all the
Steven M. Sokol wrote:
Has anyone tried installing * on a box with two eth interfaces which is
acting as a NAT box? I have only one IP at this point and I would like
to get * working without all of the NAT issues. My idea is to run * on
my gateway (which is also running the firewall and
Hello-
I've been trying to scour both * and linux user group
archives for a solution to this particular problem, but I am
just plan stuck. I got the latest zaptel sources from cvs,
uncommented ztdummy.o in Makefile, ran make; make install
then, did depmod -a. All is well up until now.
When i do
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