Gregory Junker wrote:
Good deal, I looked at the config site but wasn't sure. The SPA3K I have
for testing is 2.09, not sure if that makes a difference.
You want to upgrade. They mute the dialtone on outbound FXO calls in the
later releases.
-- PhoneBoy
__
Gregory Junker wrote:
One problem is that the SPA3K only uses two-stage dialing on the FXO -->
VoIP2 path, which means any time someone calls the phone system and gets
forwarded to a select SPA3K extension, they hear a dial tone. As far as
I can tell, there is no way to disable that. You can hav
Steve Edwards wrote:
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(helo=lists.digium.com)
by mx.perfora.net with ESMTP (Nemesis),
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Brian Capouch <[EMAIL PROTECTED]> writes:
> Rodney I don't think you're at a level of understanding of Linux yet
> where any amount of help you get on this list is going to help you.
He's right, Rodney. You really need to start at the bottom and get
comfortable with Linux before you go further.
David Mallwitz <[EMAIL PROTECTED]> writes:
> I use and highly recommend [OpenVPN].
Ditto. There's another very clear advantage to OpenVPN over IPsec,
and that's the fact that many firewalls are hard to run IPsec through,
but OpenVPN, using a single ephemeral UDP link, will work just fine.
I don
Hi!
i have to make pabx to direct calls to h323 terminals. i have an h323
gateway available and wish to use asterisk as the gatekeeper for call
direction and queueing etc.I am a beginner at asterisk and to link
openh323 with asterisk for my project i searched on net i found
different compilation
hi!
I am a beginner at Asterisk and Linux,I am trying to place a call
using IAX ,but don`t really know how to chaneg the configuration
file.I open the /etc/asterisks directory ,then open the iax.conf file
from there but can`t edit it .Can anyone please help me reagarding
this issue.How can a config
Excellent, thanks all.
Greg
Jeff Owen wrote:
I read somewhere that the 2.10 firmware fixed the hearing of digit dialing.
When I opened my unit, I upgraded it first then went to configure it so I
don't' have any real experience with the hearing the digits.
-Jeff
-Original Message-
From: [EMA
Original Message:
-
From: Luke Connolly [EMAIL PROTECTED]
Date: Fri, 19 Nov 2004 15:42:27 +1100
To: [EMAIL PROTECTED], [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Best SIP phone for high quality telemarketing
I'm really happy with my Polycom IP 600
http://www.polycom.com
A
I'd recommend reading through the included sample "queues.conf" file
carefully... what you want is easily done.
You can either do a "make samples" (which will overwrite your existing
configuration) or look in the "configs" subdirectory in the asterisk
source tree for queues.conf.sample.
Cheers
SER is a stateless and/or stateless proxy. SER by itself it not very useful
but SER teamed with Asterisk is how you make Asterisk scale. SER normally
has nothing to do with the RTP stream. Remember in the SIP world there are
really 2 parts, 1 is the SIP messages and 2 is the RTP "voice" stream.
Rodney Acosta Coya wrote:
look at this:
linux:/media/cdrom/suse/i586 # rpm -i gcc-c++*
error: Failed dependencies:
libstdc++-devel is needed by gcc-c++-3.3.3-43.25
gcc = 3.3.3-43.25 is needed by gcc-c++-3.3.3-43.25
how i can solve the second dependence
Rodney I don't think you're at
On Thu, 2004-11-18 at 15:02 -0600, Eric Wieling wrote:
> Sean Kennedy wrote:
> > Steven Critchfield wrote:
> >
> >> To the mailing list admins,
> >>
> >> The idiots running the pliva.hr mail server are not responding to
> >> messages about their broken mail server and it's insistence to send
> >>
Matthew Crocker [EMAIL PROTECTED] wrote:
> > Here's a question: if the author has purchased a commercial license to
> > use Asterisk, and I get binary modules from him, I can still use them
> > with my CVS-based Asterisk, right?
> >
> You may be able to do that. You could always run a couple Aster
I'm really happy with my Polycom IP 600
http://www.polycom.com
About $200 cheaper than cisco and no difference in qual or features.
Shaun Dawson wrote:
Hello,
I'm interested in opinions about the best phone for high quality
telemarketing use(very heavy use with highest possible quality sound).
I'd
On 18/11/2004 17:25 kido noagbodji said the following:
I just purchased 10 G729 licenses for my asterisk box from Digium I was able
to register the key. But when i start asterisk it fails with the error
message:
[codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]: loader.c:248
ast_load_resou
I've been doing a little battle with asterisk voicemail and X-Lite
(under OSX).
I can leave voicemail without a problem, everything seems to run into
problems upon checking VM.
The log is pretty descriptive:
-- Executing Ringing("SIP/xlite2-70d1", "") in new stack
-- Executing Wait("SIP
On Thu, 2004-11-18 at 14:24 -0700, Kevin P. Fleming wrote:
> Matthew Crocker wrote:
>
> > The MAXIM-921 DS from SBS supports DS3 -> DS0 channelized. It is a PMC
> > for CompactPCI it comes with Drivers for Linux, how hard would it be to
> > add libpri and asterisk support?
>
> Holy G.711 Batma
Polycom phones are nice and are about half the cost of Cisco phone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kavit
Munshi
Sent: Thursday, November 18, 2004 9:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
On Thu, 2004-11-18 at 15:45 -0600, Chase Venters wrote:
> Hello,
> I am working on a business case for Asterisk deployment, and I'm curious if
> Digium (or any other entity) has an estimated count of deployed Asterisk
> environments. I am aware of some individual deployments (Nufone, Vonage, FWD
On Fri, 2004-11-19 at 04:02, Eric Hall wrote:
> Does anyone know if you can use a Wildcard T100P with HylaFAX? I'm
> trying to setup trunking from our asterisk server to a Fax server.
You can't, the T100P is a unchannelized T1 card.
If you want to do fax, you need to look at spandsp.
How do you w
On Nov 18, 2004, at 8:23 PM, Adam Fineberg wrote:
Having some trouble with segfaults and sound quality all of a sudden (since
I recompiled from the latest source) when 2 iaxComm clients connect. First
off immediately after the server reports:
<>
<> -- Attempting native bridge of IAX2/[EMAIL PR
Does anyone know if
you can use a Wildcard T100P with HylaFAX? I'm trying to setup trunking from our
asterisk server to a Fax server.
Any help would be
great!
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Jeb Campbell wrote:
I'm replacing a Merlin for a client and they have a PagePal Intercom
that I would like to reuse.
Here is what I know about it:
It has a screw-down wires that goto rj-11 (This was told to me over the
phone) that went into one of the Merlin ports.
I tried bring it up with fxo_
On Fri, 2004-11-19 at 02:22, kido noagbodji wrote:
> hi,
>
> > That really depends on many things:
> >
> > - did you download packages or source ?
> > - if you did download source, did you have the appropiate pwlib and
> > openh323 libs installed (h323 resides in channels/h323, check it's
> > READ
I read somewhere that the 2.10 firmware fixed the hearing of digit dialing.
When I opened my unit, I upgraded it first then went to configure it so I
don't' have any real experience with the hearing the digits.
-Jeff
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
i'm running asterisk ( Asterisk CVS-v1-0-11/15/04-21:24:51 ) on a
public IP address and am accessing it with natted x-lite and spa-3000
clients. the xten client runs perfectly and i can also make and receive
calls with the spa-3000 with no issues, so both appear to be configured
correctly;
Gregory Junker [EMAIL PROTECTED] wrote:
> One problem is that the SPA3K only uses two-stage dialing on the FXO -->
> VoIP2 path, which means any time someone calls the phone system and gets
> forwarded to a select SPA3K extension, they hear a dial tone. As far as
> I can tell, there is no way to di
Good deal, I looked at the config site but wasn't sure. The SPA3K I have
for testing is 2.09, not sure if that makes a difference.
Thanks
Greg
Jeff Owen wrote:
I used http://voxilla.com message board to get mine to work. Phoneboy there
knows the spa-3k rather well and has it working with asteris
The ipo11's were 25 each when I ordered them + import costs since it
comes from TW.
Yet to use them w/ asterisk but it worked fine w/ their supplied
software in windows since they are Tigerjet based adapters.
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I used http://voxilla.com message board to get mine to work. Phoneboy there
knows the spa-3k rather well and has it working with asterisk. He created a
wizard to assist people with configuring the system to work together, give
it a look see @ http://voxilla.com/spa3kasterisk.php.
I just went thr
Can any one recommend IP phones that work the best with asterisk and
dont cause a major dent in your finances? I shall be using them for
normal office functions nothing out of the ordinary.
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http://lists.
Anybody used the above phone with asterisk
I have one working ok for calls, but having a problem with voice mail.
Using either the 'Voice mail function key' or dialing 88 (for my system)
just gets me to Call Terminated
Asterisk CLI shows the error message 'unable to get User name'
My Grandstream w
hi,
> That really depends on many things:
>
> - did you download packages or source ?
> - if you did download source, did you have the appropiate pwlib and
> openh323 libs installed (h323 resides in channels/h323, check it's
> README)
>
I installed the oh323 myself. I was just wondering if it was
Jeff Owen wrote:
I have mine working so that all incoming calls are passed directly to * and
no user heard any dial-tone or digits, even when the call goes back to the
SPA-3k for the Line1 user.
Share some config tips?
Greg
___
Asterisk-Users mailing lis
I have mine working so that all incoming calls are passed directly to * and
no user heard any dial-tone or digits, even when the call goes back to the
SPA-3k for the Line1 user.
Now when a user calls out the PSTN over the SPA-3k there is a short beep
sound and a longer delay as it takes the invite
> > While I'm not interested in the xbox part of this, I wonder how one uses
> > USB for analog connections? Explanation? Pointer to an article? Other
info?
>
> zaptel .. just as regular.
>
> There are wcusb.o modules in the zaptel drivers, that handle these. The
> usb fxs modules are part of the
On Fri, 2004-11-19 at 01:48, kido noagbodji wrote:
> Hello,
>
> I just downloaded and installed the latest version of asterisk under
> Fedora. (had it under FreeBSD but was having TOOO many problems)
> After my installation i noticed that the channel H323 was not included
> ( I remember that i di
On Thu, 2004-11-18 at 21:24, Kevin P. Fleming wrote:
> Matthew Crocker wrote:
>
> > The MAXIM-921 DS from SBS supports DS3 -> DS0 channelized. It is a PMC
> > for CompactPCI it comes with Drivers for Linux, how hard would it be to
> > add libpri and asterisk support?
>
> Holy G.711 Batman!
>
Hello,
I'm interested in opinions about the best phone for high quality
telemarketing use(very heavy use with highest possible quality sound).
I'd prefer that the phone be SIP, but I'd consider anything that will
work with Asterisk. I think I'm going to need both deskphones and
softphones, but I'
I want even less than that
All I want to do is have the SPA-3000 configured so that it offers up
its FXO and FXS ports to Asterisk -- nothing more, I want Asterisk to be
the brains.
1) The SPA should hand incoming calls on the FXO to Asterisk.
That's all I want. For an interim measure I would li
Hello,
I just downloaded and installed the latest version
of asterisk under Fedora. (had it under FreeBSD but was having TOOO many
problems)
After my installation i noticed that the channel
H323 was not included ( I remember that i did not have to install it under
freeBSD) but I have seen
Chris,
A SPARC is nice with asterisk on it but it does have limitations. I am
running a UE2 with dual proc and using Debian. However, things like IAX2
don't seem to work, there is no music-on-hold (MOH), and the MeetMe
conference server doesn't seem to work either. Those things seem to rely on
On Thu, 2004-11-18 at 17:38, Rob Emanuele wrote:
> Matthew,
>
> You should be able to use the 5 33Mhz slots easily. I'm not sure if the 2
> 66Mhz slots will work.
TE410P, works in the PowerEdge 1650 which has 66 MHz only (depending on
the PCI riser, some of them have one 66 MHz and one 33 MHz).
Steve,
There is no doubt that the SPA-3k is complicated. I used http://voxilla.com
message board to get mine to work. Phoneboy there knows the spa-3k rather
well and has it working with asterisk. He created a wizard to assist people
with configuring the system to work together, give it a look s
On Thu, 2004-11-18 at 14:35, Pascal C. Kocher wrote:
> Hello Tim
>
> I'm struggling to get a HFC card running in NT mode. It seems to work
> for a short period but then stops. The messages on the asterisk console
> mentione something about event 6.
>
> Is the zaphfc module enough to be loaded
On Thu, Nov 18, 2004 at 04:30:24PM -0600, Michael Shuler spake thusly:
> the VoIP devices and to the media gateways. The SER machines don't know
> what to do with a call they only know to hand it over to Asterisk for
> routing/CLASS features or whatever you want the call to do. You then have a
T
On Thu, 2004-11-18 at 14:25, Peter Osborne wrote:
> I had the same problem on Debian, the mpg123 in Debian is really mpg321 which
> is supposed to be a drop in replacement. Well, I don't think it is, I
> compiled mpg123-0.59r from source and it works now. You may want to give that
>
> a try.
On Thu, 2004-11-18 at 14:03, Ed Greenberg wrote:
> Over on the voip-info.org tiki I found this statement:
> > Mark (the man who made Asterisk PBX, www.asterisk.org) has an xbox that
> > has 4 analog ports via usb... aka the XBoxPBX
>
> While I'm not interested in the xbox part of this, I wonder h
Having some trouble with segfaults and sound quality all of a sudden
(since
I recompiled from the latest source) when 2 iaxComm clients connect.
First
off immediately after the server reports:
<>
<> -- Attempting native bridge of IAX2/[EMAIL PROTECTED]:4569/1 and
IAX2/4589/5
<>
<>One or both
WipeOut wrote:
Hi,
Is there any method to log the reason a call was ended / terminated /
dropped??
What do you have for busycount?
What do you have for callprogress?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html
On Thu, 2004-11-18 at 15:10, Leandro Morgado wrote:
> Thomas Jagoditsch wrote:
>
> > hi christiaan.
> >
> > Christiaan Brink schrieb:
>
> [snip]
>
> > sorry, but i use chan_capi with the avm B1 card (this is an active
> > one, different then a A1/fritzcard), see my posting.
> > i see no way to
Hello all.
Does anyone have any specific tips,hints, suggestions,etc. as to properly get
Asterisk working behind A Westell 2200 DSL router and Verizon residential DSL
service. I've been out of the Asterisk Loop for a few months but my asterisk
Service seems to be working scarce. It seems that
Hi ,
I would like to know, is there any way to interrupt musiconhold
while call in queue by entering DTMF and forward that call to
specified context ( may be voice mail etc) .
I would like to give option to people to leave voice mail if they
don't want to call for long wait time to reach age
I'm replacing a Merlin for a client and they have a PagePal Intercom
that I would like to reuse.
Here is what I know about it:
It has a screw-down wires that goto rj-11 (This was told to me over the
phone) that went into one of the Merlin ports.
I tried bring it up with fxo_ks and fxo_ls (assum
I upgraded our Asterisk from 1.0 to 1.0.2 to re-enable IAX trunking. Since then, DTMF has stopped working for certain commercial IVR's. A SIP to PSTN call is as follows:
XTEN => SER => ASTERISK => ASTERISK => PSTN via T1
The XTEN is communicating via SIP and the Asterisk boxes are communicating via
Hello,
I'm interested in opinions about the best phone for high quality
telemarketing use(very heavy use with highest possible quality sound).
I'd prefer that the phone be SIP, but I'd consider anything that will
work with Asterisk. I think I'm going to need both deskphones and
softphones, but
I must be getting thick in my old age.
The sheer number of options on the SPA-3000 is causing my eyes to glaze
over.
If anybody is willing to share their configuration I'll post a "dummy's
guide" on the wiki -- "http://www.voip-info.org/wiki-Sipura+3000"; didn't
quite do it for me.
All I want
I had the same problem when I installed Asterisk (it runs on SuSE Linux
9.0 now). You can find info in this mailing list under
http://lists.digium.com/pipermail/asterisk-users/2003-May/011185.html
(hint hint) Some Googling helps... thats how I found this link :)
--
There are 10 kinds of people in
Folks,
I am looking to find a SIP videophone that works well with asterisk
trunking in the SIP protocol. What have people used and what would you
recommend? While you're at it what would you stay away from? Pricepoint
is not a huge concern at this point but I also wouldn't mind saving my
customer
Jay Hennigan wrote:
It's the PoE that makes this switch look good to me, not the management.
That's just icing on the cake for most small office deployments.
Exactly. That's why I'd prefer a dumb 12/16-port switch with PoE, since
it would have to be cheaper than the managed version. It may only be
Yes, Asterisk supports interfacing with many different programming
languages. Basically, any program that can read from the stdin, stdout
and stderr streams can interface with asterisk. You can therefore
implement your own programs in any language that allows streaming
from/to these streams. PH
Bruce is right setting dtmfmode=inband solved the problem.
---quote---
If you are using G711, try setting dtmfmode=inband. We've had a lot of
intermittent problems with * apparently loosing or ignoring DTMF when
using rfc2833. It doesn't usually happen at the beginning of a call,
but rather aft
On Thu, Nov 18, 2004 at 10:56:26AM -0600, [EMAIL PROTECTED] wrote:
>9. TE410P - How many can I have? (Matthew Boehm)
>
> I have a Dell Poweredge 6450, 4 proc Xenon with 1Gb ram and the following
> PCI abilities:
>
> Bus type . . . . .. . three peer PCI buses: two 64-bit buses and one 32-bit
>
Hi,
I'm thinking about buying a Sun Blade 100 from Ebay. I see that it has PCI
slots. I want to run Gentoo Linux on it and install my X100P card.
My Question is...
Will the X100P card work happily with Linux on a Sparc processor?
Has anyone every tried this or the TDM400 series?
Thanks in Adv
Yes I still get the same error. Following is a section from the
compiler. I had to truncate a little
xlaw.h:1636: warning: excess elements in scalar initializer
xlaw.h:1636: warning: (near initialization for `capiINT2ALAW')
xlaw.h:1636: warning: excess elements in scalar initializer
xlaw.h:1636:
On Thu, 2004-11-18 at 12:44 -0800, Jongsuk Lee wrote:
> My guess for problem is your extension configuration file .
> You are probably detecting dtmf such as '*#' and asterisk does
> something before it sends.
> my advice is ]add those specific bank number and by pass dtmf detection stuff.
> One g
Michael Vogel wrote:
I just downloaded it. Now I only need to know, how to include it in
asterisk. The documention is ... hmm ... ;-)
http://www.e164.org/enum.phps
Little script I whipped up a while back that doesn't need anything but
the php binary to work...
--
Best regards,
Duane
http://www.
If you are using G711, try setting dtmfmode=inband. We've had a lot of
intermittent problems with * apparently loosing or ignoring DTMF when
using rfc2833. It doesn't usually happen at the beginning of a call, but
rather after a number of tones are sent, such as when picking up several
voicemail
On Thu, 2004-11-18 at 12:44 -0800, Jongsuk Lee wrote:
> My guess for problem is your extension configuration file .
> You are probably detecting dtmf such as '*#' and asterisk does
> something before it sends.
> my advice is ]add those specific bank number and by pass dtmf detection stuff.
> One g
I am in the process of writing a book on the AGI structure of * but for
now there are a couple examples on that site of how to implament it. I
learned whatI know from voip-info.org most if not everything is there
for what you may need to know.
-Original Message-
From: [EMAIL PROTECTED]
You can do what we did and setup 2 SER servers which are load balanced by a
Foundry ServerIron XL (you could use UltraMonkey for free if you prefer).
The 2 SER machines handle the REGISTER messages, NAT and final delivery to
the VoIP devices and to the media gateways. The SER machines don't know
w
Brian C. Fertig schrieb:
No nothing exists. However may I suggest PHPAGI it's a class for
asterisk to interface with it.
So something exists ;-)
You can pull channel variables etc and do all kinds of kewl junk with
it. I write all my AGI in php and execute it. But yes you in a way
can control aste
"Kevin P. Fleming" <[EMAIL PROTECTED]> wrote:
> I have yet to come across a small business office that could not be well
> served with one or two reliable 16-port switches, of the sub-$100
> variety. If they have more than 32 nodes, then they will likely have
> some on-site staff, and then a manag
Peter
- 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs
for this. If you have ISDN you can also get DDI to the extensions.I
would strongly recommend abandoning the analogue PSTN lines and using
ISDN. The setup pain you will go through will be significantly less,
combined wi
My thoughts are to have it demux'd on your end. break it into smaller
T1's and bring them in that way. Your looking at like 2-3 PRI's per box
depending on your config. This is the easiest way I could think of
getting this to happen.
.o---o
No nothing exists. However may I suggest PHPAGI it's a class for
asterisk to interface with it. You can pull channel variables etc and
do all kinds of kewl junk with it. I write all my AGI in php and
execute it. But yes you in a way can control asterisk with php at the
AGI level.
brian
Scenario A:
Lets say you had 10 Asterisk boxes, all 4U, 4 proc servers, all with same
*.conf, in a rack mount unit.
You can get 1 OC3 connection for $5,000 a month.
How can you split that OC3 among the 10 boxes and have load balancing and
auto-failover?
Scenario B:
Same setup as A, but this time,
Kevin P. Fleming wrote:
Race Vanderdecken wrote:
Why can't I convert the DS3 input to SIP Output, no transcoding,
straight G.711, all in one box?
Yes, that is what you would want to do. Probably even better would be
DS-3 to IAX, and try to get trunking support for G.711 working to keep
down
Yeah, that's definitely a very different style. In an office with 6
PCs, exactly who is ever going to run the management application for
that switch? Do any of the staff at that location even have a clue
what any of that information is for? I'd be surprised if any of my
clients even cared abou
Brian C. Fertig wrote:
I have a 3348 they don't do PoE. They do QoS and do it well. I don't
know about the upper models..
I stand corrected.
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To
Hello,
I am working on a business case for Asterisk deployment, and I'm curious if
Digium (or any other entity) has an estimated count of deployed Asterisk
environments. I am aware of some individual deployments (Nufone, Vonage, FWD,
Voicepulse).
Thanks,
Chase
--
Chase Venters
Network Engine
You also forget that the Echo Canceller will have to be hardware based to.
Its not right now.. without that the DS3 will not be possible on a PCI card
unless you do hardware echo cancel. If you do all ulaw/alaw I think you can
do a DS3 on a box :P
bkw
> -Original Message-
> From: [EMAIL
I have a 3348 they don't do PoE. They do QoS and do it well. I don't
know about the upper models..
brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Thursday, November 18, 2004 4:15 PM
To: Asterisk Users Mailing List
Does something like this exist?
Dozens of different efforts are underway along these lines.
http://www.voip-info.org/wiki-Asterisk+gui
Greg
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To UNSUBS
YES YES YES!! Boot them off! I was wondering where I was getting those
emails. Thought it was bad spam.
Matthew
- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, November 18
> > Anyways, Not That I Would Encourage Anyone To Do This, but NFR's of Netgear
> > products are available at half off list ($875 for the switch in question)
> > to Powershift partners. That's gotta be one of the better prices for that
> > switch at this time.
> >
>
> Dell has some 48 port supp
On Thu, 2004-11-18 at 16:08 -0500, Race Vanderdecken wrote:
> Okay,
>
> Dumb question.
>
> Why can't I convert the DS3 input to SIP Output, no transcoding,
> straight G.711, all in one box?
Maybe you missed the comment about the TNT boxes that already do that. I
think they may
Race Vanderdecken wrote:
Why can't I convert the DS3 input to SIP Output, no transcoding,
straight G.711, all in one box?
Yes, that is what you would want to do. Probably even better would be
DS-3 to IAX, and try to get trunking support for G.711 working to keep
down the IP overhead as we
Matthew Crocker wrote:
The MAXIM-921 DS from SBS supports DS3 -> DS0 channelized. It is a PMC
for CompactPCI it comes with Drivers for Linux, how hard would it be to
add libpri and asterisk support?
Holy G.711 Batman!
This site:
http://www.datamktg.com/modules.php?name=News&file=article&sid=53
h
Hi!
Is it possible to control Asterisk with PHP? I don't think that the
"extensions.conf" can solve all my problems. So I would like to make it
with PHP (which I really know well).
I would need a possibility to read the dialed digits and a possibility
to start a call.
Does something like this
Eric Wieling wrote:
Dell has some 48 port supposedly PoE switches for about $600. I've not
done QoS on them, but they claim to support it.
I don't see any PoE-enabled switches on Dell's web site, and the switch
you are referring to (PowerConnect 3348) definitely doesn't have any
references to 8
Sean Kennedy wrote:
Differing styles I'm thinking. Anything more than 4 clients, and I
recommend a managed switch ( in most situations ). Now, this might be
because my smaller clients tend towards growth. They aren't going to
stay small. So I set them up in a way that they will not need to w
Okay,
Dumb question.
Why can't I convert the DS3 input to SIP Output, no transcoding,
straight G.711, all in one box?
Maybe I am missing something. I am not a channel bank guy. Hate
them. I am a VOIP H.323 SIP guy.
Is there a card that will give me a netw
We are having a problem with the Polycom 300. For some reason, it will
deregister and not register back. I have looked the config files for
the Polycom, but since it is all XML I might be missing something.
Thanks.
___
Asterisk-Users mailing list
[EMAI
TC wrote:
I was told that there is $2500 PCI DS3 card available,
It must be a channelized tdm voice ds3
And not just channelized, but channelized down to DS-0. All
channelized cards I've seen only support DS-1 channels.
The MAXIM-921 DS from SBS supports DS3 -> DS0 channelized. It is a PMC
for
I read through briefly.. from what I saw no one had mentioned it.
Wouldn't this be from echotraining? What do you have echotraining set
to in zapata.conf?
On Thu, 18 Nov 2004 14:08:05 -0500, Giovanni Powell
<[EMAIL PROTECTED]> wrote:
> I tried that but still getting a delay, do you think its th
Joe Greco wrote:
Anyways, Not That I Would Encourage Anyone To Do This, but NFR's of Netgear
products are available at half off list ($875 for the switch in question)
to Powershift partners. That's gotta be one of the better prices for that
switch at this time.
Dell has some 48 port supposedly PoE
Sean Kennedy wrote:
Steven Critchfield wrote:
To the mailing list admins,
The idiots running the pliva.hr mail server are not responding to
messages about their broken mail server and it's insistence to send
everyone here a copy of a mail saying " The recipient: MARIO SPOLJAR is
no longer PLIVAs em
> Sean Kennedy wrote:
>
> > Jeeze, how can you NOT justify a 1000 bucks for a PoE switch that has QoS?
> > I was under the impression that QoS was a requirement for VoIP. Well,
> > not technically, but rationally, I wouldn't set any client up on a VoIP
> > system that didn't have a switch that
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