these are the various options i tried
[2002]
type=friend
host=dynamic
insecure=yes
dtmfmode=inband
context=from-sip
mailbox=2002
auth=plaintext
---
[2002]
type=friend
1) authenicating numbers - JT correctly pointed out, you can't allow people
to call you to verify as caller id can be spoofed. He proposed a group of
asterisk servers calling for verification. I was going to write into this
advertising info so you could get businesses to do the calling for you
Hi,
- Original Message -
From: Grzegorz Nosek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 12:52 AM
Subject: Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm,
etc.)
On Wed, 3 Dec 2003 20:45:21 +0200, Dan wrote
Hi,
- Original Message
Skuse, Phil wrote:
How do I get asterisk to populate the Calling Party Number field in an
H.323 call?
chan_h323 does not set it too ...
I have asterisk configured to accept a SIP call and connect it to an H.323
IVR system. The call goes through, but the caller id is put in the Display
field
Can you try without a pssword?
I've been running msn clients several times, and they have been working.
this was on 4.7. I'm not using it anymore, as we've bought snom and
grandstreams instead...
On Thu, 2003-12-04 at 07:54, Balaji NJL wrote:
these are the various options i tried
The library has several DSP features, including AGC, denoising, and
echo cancellation. These are all provided via integration with
preprocessing from the SPEEX library. I don't know if DAN allows you
to turn on/off echo cancellation or not. However, the echo
cancellation code from speex is
Thanks for the reply.
After a lot of digging in the oh323 code, I've discovered that if the
callerid is a valid E164 (ie. entirely composed of digits 0123456789*#) then
the callerid is put into the Calling Party Number field, otherwise the
callerid gets put into the Display field.
But there is
There are probably echo suppressors on both ends of the LD circuit
between NJ and CA, but none on the local NJ circuit.
Steve
Tom Lowe wrote:
Not a silly question. I've given that thought.
To be honest, I'm not sure what kind of phone the California or NJ
callers were using. However, we've
You need to but the license from digium $10 per
license.
And just follow the instruction attached with the
license.It will be
very easy.
--Sip Rtp
Todd Wallace wrote:
Does asterisk support G.729a or do you have to add
something (is there
an open source one)
Todd Wallace
Download the latest alsa drivers from sourceforge and make install.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Welter
Sent: Wednesday, December 03, 2003 10:27 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Soundblaster
Hi,
I have the
Paul Liew wrote:
- Original Message -
From: Paul Lambert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Paul Liew [EMAIL PROTECTED]
Sent: Wednesday, December 03, 2003 4:16 AM
Subject: Re: [Asterisk-Users] incominglimit stuck in app_queue
I've seen this same thing. But it doesn't happen
I have a SB128 (pci) and via82c686 (on board). I simply did not
compile in the drivers for both into the kernel and added
es1371
to /etc/modules. adding via82cxxx_audio below would probably give me a
/dev/dsp1 or sth :) a long time ago I had two sound blasters (awe64
pro iirc) and they worked
On Wed, Dec 03, 2003 at 10:42:40PM -0500, TeleSIP wrote:
A good rootkit will also modify the date and time of the replaced binaries
so they will look the same as the original.
Try to replace your ps command with that from a trusted RH9 machine. If
it works ok then you must do a clean
On Thursday 04 December 2003 08:27 am, PJ Welsh wrote:
On Wed, Dec 03, 2003 at 10:42:40PM -0500, TeleSIP wrote:
A good rootkit will also modify the date and time of the replaced binaries
so they will look the same as the original.
Try to replace your ps command with that from a trusted
-- Original Message --
From: Tim Thompson [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
Date: Wed, 3 Dec 2003 13:22:25 -0600
I would change the option number to something else because 9 is often
picked up in another context as 9NXXNX
You might have
Aaah yes,
I experienced similar problems. I had problems making calls to/from
calls through my cards. I had to play with the echo cancellation to get
it to manageable levels.
If this is going to be more of a mainstay installation, I would highly
recommend that you get a T100P card and
On Dec 3, 2003, at 8:22 PM, Peter Zeltins wrote:
The library has several DSP features, including AGC, denoising, and
echo cancellation. These are all provided via integration with
preprocessing from the SPEEX library. I don't know if DAN allows you
to turn on/off echo cancellation or not.
i'd like to be able to set up an extension, that when it rings, it will
dial the console, have it auto answer and play my sound file.
what im trying to accomplish is to use my paging system for an external
ringer as well. im pretty sure i can get the other parts set up if just
knew how to get
Hello everyone:
I've been lurking on this list for a bit now and reading about *. The
project seems to have great momentum and could-always-be-better
documentation. I can relate! :^)
We would like to get some help from the experts on this list getting a
prototype installation installed at
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to the
Call Manager I released last week) and it seems to regularly miss
incoming calls entirely. It also occasionally misses the drop signal
Same here.
Hi Scott,
Similar to your experience, I've been pretty happy running Asterisk under
Fedora. My server has a single Xeon CPU and seems to run circles around Red
Hat 8.
Regards,
Jeff Gunther
Intalgent Technologies
[EMAIL PROTECTED] wrote on 12/04/2003 10:20:37 AM:
Hi all-
Over the past
Hi all-
Over the past week or two, I've been trying out asterisk under Fedora 1
Linux (RedHat). In my setup (single and dual Xeon motherboards), I have so
far had a very good experience in terms of performance. In doing E1 load
testing, I've found that Fedora handles heavy load much better than
I need to know if someone encounters display errors (like the window
displayed partially) when some 'strage' resolutions are used for the
display
in Windows XP native theme mode.
I am running XP in 1280 x 1024 x 2 Monitors. The bottom status window
is cut off in DIAX 0.9.5.
Also: I finally
i use google, with site:digium.com to search the archives, but i've never
found a way to show the newest messages first, or limit the results to
messages within a date range. anybody know a better way to search that
allows this?
___
Asterisk-Users
What kernel version and what patches does Fedora come with?
On Thu, 2003-12-04 at 16:43, [EMAIL PROTECTED] wrote:
Hi Scott,
Similar to your experience, I've been pretty happy running Asterisk under
Fedora. My server has a single Xeon CPU and seems to run circles around Red
Hat 8.
I've only been using Asterisk with Fedora for a short time now, but I have
had no trouble with it on my older server and a T1 card from digium. My
server is a Pentium III 733 MHz with 512Megs of RAM. Nothing special, but
gets the job done. The only problems I've heard from people with Redhat's
Is there a setting on the meetme room to shorten the delay. When someone
speaks, there is a long delay until the sound is actually heard?
Todd Wallace
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I get 2 ringing sounds when placing a SIP call through my carrier. the
first sounds European for 1 ring then, it goes to a US ring.
Any thoughts?
Todd Wallace
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Yes, I also had a problem with RedHat's threading issue (zombie AGI
processes), under Redhat 9 too, but this can be gotten around by specifying
the following before starting asterisk:
export LD_ASSUME_KERNEL=2.4.1
Thanks,
Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Email:
I don't know if anyone had used these boxes yet.
I've installed some, but not connecting to an * system. The guys who
ordered them said they ran about $600.
They are pretty cool in that you designate what the port will be either
Data, FXS, or FXO(not fully implemented yet)
This is a NEW price
2.4.22
see also: fedora.redhat.com
cheers.
Scott M. Stingel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roy Sigurd Karlsbakk
Sent: Thursday, December 04, 2003 3:57 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] Experiences
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the default
apache installation that comes with Redhat.
Here is what I get in my error logs:
[Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism
enabled (wrapper: /usr/sbin/suexec)
[Thu Dec 04 11:59:58
Steve,
You really should bring this up on the iaxclient-devel list, either
instead of or in addition to here.
I'm no longer maintaining the IAX1 compile, so there may be new
features added to the library (such as the handling of text frames, as
you've noted below) which will cause the IAX1
- Original Message -
From: Todd Wallace [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 11:36 AM
Subject: [Asterisk-Users] ringing
I get 2 ringing sounds when placing a SIP call through my carrier. the
first sounds European for 1 ring then, it goes to a US
Dear Rob,
I'm a solo Asterisk vendor in Philadelphia, PA (www.tlsolutions.net). I
would like to submit a reply to your RFP. I would also like you to consider
other hardware recommendations for your * systems.
Are you available by telephone to discuss your project in more detail?
Regards,
On Thu, 4 Dec 2003, Scott Stingel wrote:
Hi all-
Over the past week or two, I've been trying out asterisk under Fedora 1
Linux (RedHat). In my setup (single and dual Xeon motherboards), I have so
far had a very good experience in terms of performance. In doing E1 load
testing, I've found
Is there a wait or a setting that I can set so that * does not do this?
It sounds like you're receiving ringback from your local asterisk first.
Then, somewhere along the progress, your asterisk receives an open channel
and connects you to the sip carrier. At this point, the carrier's channel
- Original Message -
From: Todd Wallace [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 1:04 PM
Subject: Re: [Asterisk-Users] ringing
It sounds like you're receiving ringback from your local asterisk first.
Then, somewhere along the progress, your asterisk
jerk face wrote:
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the default
apache installation that comes with Redhat.
Here is what I get in my error logs:
[Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism
enabled (wrapper: /usr/sbin/suexec)
[Thu
I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel
bank (12fxs/12fxo). I have the setup partially working thanks to some help from
IRC. However I still have the following issues I can't seem to resolve
1. When calling into the system from the PSTN call hangup is not
On Thu, 4 Dec 2003, Olle E. Johansson wrote:
jerk face wrote:
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the default
apache installation that comes with Redhat.
Here is what I get in my error logs:
[Thu Dec 04 11:59:57 2003]
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
jerk face wrote:
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the
default
apache installation that comes with Redhat.
Here is what I get in my error logs:
[Thu Dec 04 11:59:57 2003]
RedHat's PERL doesn't allow suid. You'll have to turn of the s flag
on vmail.cgi (chmod -s /var/www/cgi-bin/vmail.cgi) and fiddle with
permissions.
-wade
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, December 04, 2003
Arnold Ligtvoet wrote:
Leif wrote:
Awesome! Have you tried the newer patch / diff for 1.259 (which as of
right now is the newest chan_sip file). If you goto bugs.digium.com and
login anonymously and jump to bug 104, then you can get the newest
patch. Same instructions as
--- Lists [EMAIL PROTECTED] wrote:
On Thu, 4 Dec 2003, Olle E. Johansson wrote:
jerk face wrote:
I recently switched from Mandrake to Redhat and
I
noticed that vmail.cgi does not work with the
default
apache installation that comes with Redhat.
Here is what I get in my error
On Thu, Dec 04, 2003 at 07:56:56PM +0100, robert ivanc wrote:
this patch seems to break my GS phones that are connecting to * via NAT.
The one before that works ok - 249 or something? They can't connect
anymore - get a Not Found error back.
That is very strange -- the *only* difference
--- Wade J. Weppler [EMAIL PROTECTED] wrote:
RedHat's PERL doesn't allow suid. You'll have to
turn of the s flag
on vmail.cgi (chmod -s /var/www/cgi-bin/vmail.cgi)
and fiddle with
permissions.
-wade
Once I turn off the 's' flag, I can run the program
but I can't view the messages. By
Is it possible to initiate 2 outbound calls from a web page and conference
them together in a bridge on an asterisk server?
Todd Wallace
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Here is a script I use in a cron job that runs every 5 minutes to make it so
that my webserver (which runs as the apache group) can access the voicemails
through the web. Seems to fix my problems. Although if I get the email
there is a voicemail it might be 5 minutes before I can get to it via
-Original Message-
From: jerk face [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 9:02 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] vmail.cgi with Redhat 9.0
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the default
On Thu, 2003-12-04 at 09:30, [EMAIL PROTECTED] wrote:
i use google, with site:digium.com to search the archives, but i've never
found a way to show the newest messages first, or limit the results to
messages within a date range. anybody know a better way to search that
allows this?
Thats
On Thu, 2003-12-04 at 13:21, Todd Wallace wrote:
Is it possible to initiate 2 outbound calls from a web page and conference
them together in a bridge on an asterisk server?
Yes, sample.call. The first part is the phone number to dial, and the
application is dial with the other phone number.
--
Changes are below. Use KewlStart for the FXO channels. (Loopstart +
remote disconnect suppervision) Define all T1 channels. FXS channels
can be loopstart without any issues.
quote who=Jonathan Moore
I just purchased a T100p from digium and a Carrier Access Access Bank 1
channel bank
On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel
bank (12fxs/12fxo). I have the setup partially working thanks to some help from
IRC. However I still have the following issues I can't seem to resolve
1. When
While I have not had any dealings with this company, I really enjoy that
you can have open and transparent dealings here. I have seen this of
one or two other companies on other lists and find it refreshing to hear
about how companies are doing especially when they help the community.
On Wed,
Guys, I'm using RH9 with vmail.cgi without any modifications... I'm just do
a 'make webvmail' after 'make install'... I don't have any troubles...
Regards,
Gus
- Original Message -
From: Carlton J. O'Riley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 4:23 PM
Obviously, there are no DS3 TDM cards that are currently compatible
with Zap channels. (or are there?)
Does anyone know of an inexpensive DS3 card that could perhaps be
used with Asterisk if one were to try to port the Zap drivers to such
a card? PCI, of course, would be the bus of choice.
On Thu, 2003-12-04 at 09:30, [EMAIL PROTECTED] wrote:
i use google, with site:digium.com to search the archives, but i've
never
found a way to show the newest messages first, or limit the results to
messages within a date range. anybody know a better way to search that
allows this?
Thats
I've patched app_voicemail.c to
create everything as 777. As this is a
dedicated Asterisk box, I don't see
the harm in giving everyone on the
system full access.
Would you be willing to share this patch?
It was a simple patch. Just search for 0700 in app_voicemail.c and
change them all to
What's the correct way to do cvs update now?
'cvs update' seems to work in the asterisk directory, but not the zapata
or other source directories.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
--- Evan P. Hall [EMAIL PROTECTED] wrote:
-Original Message-
From: jerk face [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 9:02 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] vmail.cgi with Redhat 9.0
I recently switched from Mandrake to Redhat and I
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
I just purchased a T100p from digium and a Carrier Access Access Bank 1
channel
bank (12fxs/12fxo). I have the setup partially working thanks to some help
from
IRC. However I still have the
Thanks for the help. Can you explain the need to define all the channels in
zapata.conf? I am not connecting devices to all the ports on the CB yet, so if I
place the definitions into my groups 1 and 2 then things seem to be a bit
strange when defining my outbound pstn calling.
Quoting Robert
I believe there are boxes that will take a DS-3 from the Telco and spit
out T-1's to your telecom equipment. Not sure what they are called.
John Todd wrote:
Obviously, there are no DS3 TDM cards that are currently compatible with
Zap channels. (or are there?)
Does anyone know of an
On Thu, 2003-12-04 at 14:06, John Todd wrote:
Obviously, there are no DS3 TDM cards that are currently compatible
with Zap channels. (or are there?)
Does anyone know of an inexpensive DS3 card that could perhaps be
used with Asterisk if one were to try to port the Zap drivers to such
a
I have been in contact with OnTrack Studios and he male voice work for
asterisk. If you wish to contact him [EMAIL PROTECTED]
I know someone on the list was looking for a male voice.
Thanks,
Brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Thu, 2003-12-04 at 14:09, [EMAIL PROTECTED] wrote:
On Thu, 2003-12-04 at 09:30, [EMAIL PROTECTED] wrote:
i use google, with site:digium.com to search the archives, but i've
never
found a way to show the newest messages first, or limit the results to
messages within a date range.
If anyone is interested, I've trimmed one of Allison's recordings down
to the single word 'welcome', for use as a generic first message when a
line is answered.
I've put it up at:
http://jhcloos.com/sounds/asterisk/welcome.gsm
and will submit it to bugs.digium.com as well.
-JimC
On Thu, 2003-12-04 at 13:42, Jonathan Moore wrote:
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 11:49, Jonathan Moore wrote:
I just purchased a T100p from digium and a Carrier Access Access Bank 1
channel
bank (12fxs/12fxo). I have the setup partially working
http://www.mail-archive.com/asterisk-users%40lists.digium.com/index.html
Returns searches in chronological order.
___
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all,
I've got some questions to post in regard to running asterisk in a
production-grade environment, specifically targeting high-density IVR
applications. No VoIP involved, just straight PSTN - * and perhaps the
occasional outdials or agent-based predictive dialing.
1) Which user would you
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 13:42, Jonathan Moore wrote:
Quoting Steven Critchfield [EMAIL PROTECTED]:
On Thu, 2003-12-04 at 11:49, Jonathan
On Thu, 2003-12-04 at 15:29, Ahmad Faiz wrote:
Hi all,
I've got some questions to post in regard to running asterisk in a
production-grade environment, specifically targeting high-density IVR
applications. No VoIP involved, just straight PSTN - * and perhaps the
occasional outdials or
Answering question number 5 only:
My customer's system is an extremely busy IVR, used in a game-show call-in
environment, with short calls and high peak call rates. The maximum number
of ports so far that my system can handle, with a single fast P4 processor,
is 4 E1 spans (one E400P). Even at
Don't have answers to your main questions but there is a place share war
stories. The Asterisk Wiki
http://www.voip-info.org/wiki-Asterisk
Not that many scenarios posted, but a few.
--
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508
Quoting
I don't want to criticize your idea, but you do have to consider certain
points. Starting from (as has already been mentioned) the bandwidth of DS3
is far too much to reasonably shove down the PCI bus without data loss /
excessive overheads. Thus a sensible approach would be one where the card
A customer contacted us today concerning getting a VoIP to PSTN system with
a few IP Phones setup. Asterisk should fit his needs. It is not a big job,
but I think that this customer is going to need onsite work.
Please contact me off list if you are an interested reseller in the
Washington, DC
At 02:34 PM 12/4/2003, you wrote:
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be generated per month on a
DS3?
Eight quad-span T-1 cards from Digium:
Correcting an idiot-math error (24/4 != 8 and 1000*3 != 1000) ...
At 02:34 PM 12/4/2003, you wrote:
However, considering the traffic volumes that you are talking about, is it
really true to say that the traditional telco cards are astronomically
priced, given the amount of revenue that can be
On Thu, 2003-12-04 at 16:52, William Waites wrote:
On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote:
I don't want to criticize your idea, but you do have to consider certain
points. Starting from (as has already been mentioned) the bandwidth of DS3
is far too much to reasonably
Title: Asterisk and Avaya IP phones
The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office.
Question: Can I make this IP telephone register and work with
I am uncertain of PCI bus speed limits - too many conflicting reports
are wedged into my head.
However, the intent here is to dump calls out via VoIP and not simply
switch between channels elsewhere on the DS3, so overcoming that
limitation needs to be addressed (if it exists at all, as a
I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000 would be a much better platform, but I don't know if you can
find one used.
Stephen
-Original Message-
From: Ernest W. Lessenger
Lucent TNT box price is attractive, but based on real experience it is
not very VOIP friendly. You have to consider it. It is hard to
interconnect with Cisco for example. I have no idea about Max
TNT-Asterisk interconnection.
We are using Nextone softswitch and able to serve clients and
On Thu, Dec 04, 2003 at 04:58:03PM -0600, Steven Critchfield wrote:
a standard 32 bit 33MHz PCI bus has a maximum bandwidth of
133MBps == 1Gbps. a DS3 is 45Mbps. even if you pass the data
over the bus 10 times, you're still only using up half the
peak bandwidth.
Thats only if you
There are DS3 (and OC-3) PCI cards available
with Linux drivers (for data). Might be worthwhile
contacting a vendor of those things to see if there's
a way to suck the TDM voice data
off a channelized DS3.
___
Asterisk-Users mailing list
[EMAIL
Ok, I contacted the seller about the ring issue. He has offered to replace the
fxs card in the unit.
1. Is the ring generator on the fxs card or part of the chasis?
2. Can anyone confirm the appropriate jumper settings for connecting analog
phones to CB?
--
Jonathan Moore
Director of
Greg Boehnlein wrote:
First and foremost, these Key System installers are big believers
in VoIP and convergence technologies. While the KSU vendors may see
This has been my experiance as well. Everybody but PBX vendors like
VoIP. The KSU people like it because it gives them more work and
On Thu, Dec 04, 2003 at 03:31:06PM -0800, David Boreham wrote:
There are DS3 (and OC-3) PCI cards available
with Linux drivers (for data). Might be worthwhile
contacting a vendor of those things to see if there's
a way to suck the TDM voice data
off a channelized DS3.
I know of OC3 ATM
I haven't personally switched to Fedora but I did decide to upgrade a lot
of the packages on my * box from RH9 to Fedora. I have not spent a lot of
time monitoring how it has handled the load but it does seem to run quite
smoothely. After having installed many of the packages to satisfy
On Thu, Dec 04, 2003 at 06:25:16PM -0500, William Waites wrote:
btw, jason thorpe at nasa has benchmarked gige cards on netbsd/i386
doing well in excess of 500Mbps so it /is/ possible.
Just another data point:
We also made measurements in November 2000 from a Pentium III running
Linux
On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote:
What's the correct way to do cvs update now?
'cvs update' seems to work in the asterisk directory, but not the zapata
or other source directories.
I use 'cvs update -PAd'
AFAIK it should work in the zapata and libpri
Anton Yurchenko wrote:
Michiel Betel wrote:
Anton,
Take a look at the latest version of the patch in:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214
It does adds an abiliti to make an announcment to a user once they are
in queue, but no this behaviour with cheking if all
John Todd wrote:
Obviously, there are no DS3 TDM cards that are currently compatible
with Zap channels. (or are there?)
Does anyone know of an inexpensive DS3 card that could perhaps be used
with Asterisk if one were to try to port the Zap drivers to such a
card? PCI, of course, would be
On Thu, 4 Dec 2003, John Todd wrote:
To Steven's comments: Yes, I have considered multiple Asterisk
devices and I am very aware of de-muxing DS3's into individual T1's
or PRI's (which bring it's own set of problems, since there is no
multi-PRI D-channel support in * at the moment)
Ahh..
Hi All.
I'm working on an * configuration. We require 8 inbound POTS lines, and
CT1 or PRI seems like it will be
quite expensive at that level. I've read that a T1 Channelbank plus
the T100P would be a (the?) way to go
for this situation. What is the recommended channelbank for use in this
Title: Asterisk and Avaya IP phones
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Ed
RubrightSent: Thursday, December 04, 2003 5:03 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk
and Avaya IP phones
The company I work for
On Thu, Dec 04, 2003 at 06:26:49PM -0600, Paul Oster wrote:
Hi All.
I'm working on an * configuration. We require 8 inbound POTS lines, and
CT1 or PRI seems like it will be
We have an installation with 9 inbound voice channels (one is the fax) and 768K data.
It is a Hybrid PRI. It
I just went through this cycle. I did go ebay and went with a Carrier Access
Access Bank I, since I still haven't decided if this will be our production
system and I didn't want to invest too much to find out. After fighting with it
all day today, I would not recommend the AB1 for this
You don't need zapata anymore...
'make update' works in asterisk directory.
bkw
On Thu, 4 Dec 2003, William Waites wrote:
On Thu, Dec 04, 2003 at 02:20:20PM -0600, Rich Adamson wrote:
What's the correct way to do cvs update now?
'cvs update' seems to work in the asterisk directory, but
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