When your phones acces to voicemail or to an IVR into the asterisk then
a G729 license is used so Asterisk is transcoding. So you are gonna use
the licenses for sure, maybe not from phone to phone calls, but yes
using the Asterisk functions.
Adrià Vidal
In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> I'd recommend the VIA EPIA PD series of boards. If you want to be safe
> load-wise get the PD 1 which has a C3-2 (Nehamiah core) processor at
> 1Ghz. The board has two ethernet ports and one PCI slot. Combine this
> with Astlinux
Hi, does anybody have a working config or tips to connect the welltech wellgate 3804 (4fxo) unit to asterisk via SIP ?I think I register it via SIP with my * box, but when sending calls from * to the wellgate the unit does not pass the call to any of the fxo ports.
thanks in advance,--
On Mar 22, 2006, at 8:25 PM, Nathan Alberti wrote:
I hope this isn't considered cross posting, i sent the following email
to Digium support but figured someone on the list may also have better
insight into my questions.
I have purchased 2 g729 licenses from Digium for testing and have the
On 03/23/06 03:08 Mojo with Horan & Company, LLC said the following:
Poor Andrew, everyone just comments how cool his email is ;)
I think the problem is:
exten => 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
should be
exten => 2,3,Set(DB(forward/${CALLERIDNUM}) = ${FORWARD})
Note
On 03/23/06 02:17 Erik Anderson said the following:
On 3/22/06, Andrew D Kirch <[EMAIL PROTECTED]> wrote:
Andrew D Kirch
Indianapolis, United States
Well if that isn't one of the most bizarre emails I've seen come
across this list.
but hey, it did make me laugh ! :)
--
Regards,
Dear sir,
I got trouble on InnoMedia video phone with
Asterisk-1.2.4.If InnoMedia video phone as a caller, then the call will be a
success, no any problem.
The problem happens:If InnoMedia video phone to be a
callee, the call can not make successfully.For instance, Caller 23267668
dialed
Brian Capouch wrote:
Martin Joseph wrote:
On Mar 22, 2006, at 2:49 PM, Avi Miller wrote:
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my exper
I'm not exactly sure the link to get to it, but I know the SIP firmware is
located in the NON-SIP firmware section on cisco's site... It's a few
links under the SCCP version.
The SIP version firmware for the 7970 SUCKS as far as I can tell (what we
tried anyway... ended up setting it up using
Greetings Asterisk Users,
AstriCon (the Asterisk Conference) is on its way to Europe. AstriCon
will be held in three cities this coming summer: Berlin, Paris and
London. Early bird registration is now open. Early bird participants
save 20% off the standard price of the events. Early bird
regis
I hope this isn't considered cross posting, i sent the following
email to Digium support but figured someone on the list may also have
better insight into my questions.
I have purchased 2 g729 licenses from Digium for testing and have the
following questions;
** My configuration is a sin
Hello!
I'm hearing about this 7970 SIP firmware. I'm a Cisco Registered
Partner with full access to the Cisco Software Center, and yet I
can't find it. Can someone enlighten me as to where to get it?
Is it also available/applicable to the 7971G-GE?
Did you know that on March 10, SIP 8.2 was
I really don't this is a phone issue.
From: [EMAIL PROTECTED] on behalf of Douglas Garstang
Sent: Wed 3/22/2006 11:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
Asterisk-Users@lists.digium.com
Subject: RE: [Asterisk-Users] polycom queue bug
Sounds like an implementation bug. I have not seen this issue with Polycom 601
phones and Asterisk queues. I'd suggest you get ngrep or tethereal and run it
on your Asterisk system (port 5060) to see who's sending what to where.
Doug
-Original Message-
From: [EMAIL P
I was having trouble getting hints to work with my GXP-2000 (with the beta
firmware). I am running Asterisk 1.2.5. I had hyphens in the SIP channel
names and it wasn't working. I have changed them to underscores and it has
worked in 1.2.5. So I would say that it is not yet fixed in Asterisk <=1.2.5
Andrew Furey wrote:
Not technically true, AFAIK... the reverse doesn't have to be the same
(how would multiple domain hosting work?) but it should resolve to
_something_ (NXDOMAIN is not an option).
Yes, true. :)
--
National Manager - Special Projects
< Sydney / Melbourne / Canberra / Hobart
Hi Michael --
There is a hardcoded limit of 7 buddies that the Polycom IP phones support
with the current firmware. Polycom is rumoured to be increasing this limit
to 42 in a new version of the firmware due for release next quarter.
Mike.
-Original Message-
From: [EMAIL PROTECTED]
[mailt
Kyle Sexton wrote:
TE406P:
- PRIs will go from up and working fine, to "Provisioned, Down, Active"
after the server has been up for around 10 minutes, this may be
related to
rxfax and txfax being installed? Has anyone had an issue with this
specific
card? We have had this experience across
On 3/23/06, Avi Miller <[EMAIL PROTECTED]> wrote:
> Not only does 'host.domain.com' need to resolve to an IP
> address, but that IP address must resolve to 'host.domain.com' in the
> reverse lookup table.
Not technically true, AFAIK... the reverse doesn't have to be the same
(how would multiple do
Kyle Sexton wrote:
TE410P:
- zttest will never report 100% for me across different motherboards
(Supermicro P8SCT, Dell 850)
- Crash/instability of about once per two weeks where I have to power cycle
the server, i.e. phone calls stop working and a reboot fixes it
TE406P:
- zttest runs flawles
hugolivude wrote:
what's a smart host?
A smarthost is another SMTP server (e.g. your corporate email server,
which should already be capable of sending outbound email) that your
Asterisk box is configured to send all outgoing mail to, instead of
trying to deliver it directly.
The smarthost
what's a smart host?
On 3/22/06, Avi Miller <[EMAIL PROTECTED]> wrote:
> hugolivude wrote:
> > I believe that part of my problem is the fact that my static, external
> > IP address is not mapped to a hostname.
>
> I think its the reverse lookup as well, that's the source of your
> problems. Not on
I'm having a problem with polycom ip601.
If I Dial() directly eg Dial(SIP/4000) it works perfectly. The polycom
rings, and stops ringing as soon as I hang up.
But if the phone is called via a queue, the polycom continues to ring long
after I've hung up.
Other phones in the queue (grandstrea
what's a smart host?
On 3/22/06, Avi Miller <[EMAIL PROTECTED]> wrote:
> hugolivude wrote:
> > I believe that part of my problem is the fact that my static, external
> > IP address is not mapped to a hostname.
>
> I think its the reverse lookup as well, that's the source of your
> problems. Not on
that means chan_zap is not communicating well with zaptel driver. I
have never had that problem with zapata, wich version of asteris are
you using? Post more relevant configuration of zapata.conf, like the
channels configured, does ztcfg gives any errors?
Regards
On 3/22/06, Mimmus <[EMAIL PROTEC
hugolivude wrote:
I believe that part of my problem is the fact that my static, external
IP address is not mapped to a hostname.
I think its the reverse lookup as well, that's the source of your
problems. Not only does 'host.domain.com' need to resolve to an IP
address, but that IP address
On Thu, 23 Mar 2006, john wrote:
Hi,
Does anyone know how to define speeddials in XML for the 7970 sip firmware?.
I've played with the SEP.cnf.xml file that was posted previously but
can't find a way to do it. I can define them on the phone usually (seems a
bit buggy) but if the phone reboot
First off, thanks to all for your help. I'm still not operational, so
I've decided that I'll have to _understand_ what's going on rather
than relying on a magic recipe; which isn't a bad thing I guess!
I believe that part of my problem is the fact that my static, external
IP address is not mapped
I have a problem with my Polycom phones. In the buddy list, the phone
displays all but three employees.
For those three employees, there is no difference in any of the
configurations.
Is there a secret to getting all employees into the buddy list?
Thanks,
--
Michael Welter
Telecom Matters
On Thu, 2006-03-23 at 02:17 +, john wrote:
> Hi,
> Does anyone know how to define speeddials in XML for the 7970 sip
> firmware?. I've played with the SEP.cnf.xml file that was posted
> previously but can't find a way to do it. I can define them on the
> phone usually (seems a bit buggy) but
Hi,
Does anyone know how to define speeddials in XML for the 7970 sip
firmware?. I've played with the SEP.cnf.xml file that was posted
previously but can't find a way to do it. I can define them on the
phone usually (seems a bit buggy) but if the phone reboots they get lost
from the config. Do
On 3/22/06, patty McHenry <[EMAIL PROTECTED]> wrote:
>
> Do you remember when Digium annouced channelized DS3???...vapour. You watch,
> this CODEC board will go the way of the DS3 boards...vapour.
>
> Mark my words.
>
I saw the board personally last week at VON. It is far from vaporware.
--
Bir
On Wed, 22 Mar 2006, Martin Joseph wrote:
On Mar 22, 2006, at 10:32 AM, Andrew Latham wrote:
I think you would need to alter the firmware to set the kewlstart to
FXS instead of FXO. This is just a thought, I have not done such. I
decided that if it worked then such a device would have been marke
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
We will be able to do that using ASTPP whenever I have time to spend
on Local channel problems that have cropped up for me. That is part
of the reason that I have support builtin for the local channel. I
guess it doesn't actually drop them back but i
Michael Collins wrote:
> Just curious if someone out there might have already solved this problem
> and created a Python module that you could borrow...
A python package is available from
http://py-asterisk.berlios.de/py-asterisk.php. It doesn't seem to be
activly maintained but it might serve as
Do you remember when Digium annouced channelized DS3???...vapour. You watch, this CODEC board will go the way of the DS3 boards...vapour. Mark my words.
Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates.___
--Bandwidth and
Hi Greg,
Our experience is that both Asterisk and Nortel are capable of
understanding DMS100 enough to each be able to connect to a real DMS100
- however neither is capable of actually being a DMS100.
We actually ended up using 2 PRIs between our Nortel 11C and Asterisk -
the first is set up
Hadley Rich wrote:
Apologies, yes I was going mad. It just goes to show that you should always
check everything possible, even the simple things -- it was the cable. Odd
since it worked with the X100 and another TDM card but there you go.
I had a similar experience: One of my users kept report
Hi Will,
You should use the standard rfc2833 as
suggested.
Also be aware that using "ulaw" will always switch
to "inband" DTMF mode, but the phone and Asterisk should automagically
detect this.
Btw: Cellphones and DTMF just doesn't work well
together.
If the cellphone is in a poor recept
On Thursday 23 March 2006 09:38, Hadley Rich wrote:
> Is anyone else having this problem or am I just going mad?
>
> FWIW I just tried an old X100P on the line and it works correctly.
>
> I don't think I am doing anything wrong in my configuration.
OK, self reply again.
Apologies, yes I was going
On 3/22/06, Dr. Michael J. Chudobiak <[EMAIL PROTECTED]> wrote:
> Hadley Rich wrote:
> > Hi all,
> >
> > I have hit a wall configuring a TDM400, I have set these up before without
> > issue but today I just can't seem to figure out what I am doing wrong.
>
>
> I couldn't make TDM400/FXO work on my
That is will be a lot of work.
First comment I would make is the aastra's will give you issues. I have
a shop with 30 of them and there were very problematic and a poor
speaker phone. They are swapping all of them to polycoms.
My next question would be what are you going to use for PRI hardwar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Thursday, 23 March 2006 5:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VERY IMPORTANT(TREAT WITH URGENCY)
Erik Anderson wrote:
> On 3/
On Mar 22, 2006, at 2:49 PM, Avi Miller wrote:
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my experience.
That's what I was thinking also. In a
Bob McDowell wrote:
I personally wish you all the luck in the world. Any healthcare
organization looking to save like this is a worthy one.
Ditto. Plus the check everything twice before demo'ing advice. :)
My shop is small. Just 16 lines, 60 or so sets, and 3 DID's (actually
being used.)
That's definitely an undertaking, I would suggest taking it a section at a
time. We're currently running about 61 phones on our system, about to
ramp up to about 1300 by mid summer, and then to about 10,000 by next
summer. It's definitely not something you want to do in one fell swoop,
and I
Hello all,
I have Asterisk 1.2.1 and a TE110P connected to a Nortel Meridian Option
81C system. The PRI line is currently setup as DMS100. Here are the
relevant lines from zaptel.conf and zapata.conf:
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone= us
defaultzone = us
Title: Asterisk Users
This is beyond obvious, but make sure you test
everything - Every Single Thing - right before you go in there. I have had
a few instances where I reach for a nifty feature and discover that I broke
it yesterday when I implemented 'X'. This happened in a mini-demo of mine
It's ok, no offense taken. for real, I shouldn't have even wasted
everyone's bytes replying to the spammer, but I thought it was too
funny. Like a koan. Telling someone I speak only english but telling
them in another language... Shouldn't even be wasting these bytes now ;)
Leo Ann Boon wrot
Steven wrote:
Are you happy with freePBX?
Very happy. Anything I can't do via the web interface, I can do in the
*_custom.conf files. Also, with the new module system, the ability to
extend freePBX is even easier. :)
--
National Manager - Special Projects
< Sydney / Melbourne / Canberra /
You will most likely need an E1 crossover
cable.
With regards to the dialplan programming, E1
connections (internal and external) are much the same - so dial(ZAP/G1) and so
on will work fine.
regards,
Paul HalesTechnical
ManagerAsteriskIT
- Original Message -
From:
r
We are in final testing and will shortly begin shipping IAX CPE devices with
the following configurations:
1 FXS
2 FXS
1 FXS, 1FXO
These devices have an integrated gateway router with IPSec VPN, QoS, and
more.
Codec support includes G.711, G.723.1, G726, and G 729a/b.
Please contact me off-lis
Mojo with Horan & Company, LLC wrote:
lol mine was the reply not the rant ;P
translate it to find something to the effect of 'this is an english
list. That's all I speak.'
My apologies.
___
--Bandwidth and Colocation provided by Easynews.com -
Hi ..., I've a
TE205P card installed in my asterisk box.Port 1 of my card is connected to
service provider.From port 2 I want to connect Avay Partner system.what type
of cable I require to connect the partner system (straight/cross over). How the
call routing from outside will be done to
Thanks,
I just started my download of 4.3
Hopefully, I can get most of it build this week.
Are you happy with freePBX?
I originally started using asterisk via [EMAIL PROTECTED] , but found using a
database limiting.
But at the time [EMAIL PROTECTED] was relatively new and any asterisk
referen
> Is there anyway I can make one particular extension always dial out on
> one specific pots line(group)
Set the context= for the extension's sip/iax/zapata.conf entry and then in
that context dial out the specific Zap channel/group.
Nabeel
___
--Bandw
Title: Asterisk Users
Can you guys and girls give me some examples of companies using Asterisk and how many DIDs you have. I have built a small system and tested it with AASTRA 480i's and all is working perfectly. I go in front of my Management Board tomorrow to demo the app and show them i
Will Glass-Husain wrote:
My local phone is a Grandstream GXP-200
dtmfmode=info
For the GXP2000's, you want to change this to:
dtmfmode=rfc2833
They don't really handle INFO mode well, in my experience.
--
National Manager - Special Projects
< Sydney / Melbourne / Canberra / Hobart / London
Hi,I'm struggling a bit with DTMF. It seems to work fine on my internal network, but when I call outside lines with telephone trees, some systems understand the DTMF and some ignore it. Anyone have tips on solving this? Thanks in advance.
My local phone is a Grandstream GXP-200mailbox=89username
On 3/22/06, Andrew D Kirch <[EMAIL PROTECTED]> wrote:
exten => s,2,GotoIf($[${CALLERIDNUM}<300]?s,5) ;since 1xx is the pattern match for internal extensions anything lessthan 300 has to be internal so we already know that that is theextension they are wanting to forward
Downright hilarious.
Steven wrote:
Has anyone built a stable server with 64bit Centos?
If so, which Kernel?
Yup. I have Asterisk 1.2.5 with freePBX 2.0.1 running on CentOS 4.3
x86_64 (2.6.9-34.ELsmp). Note that CentOS 4.3 has the same spinlock.h
bug as RHEL4 Update 3, so you'll need to take that into account when
When you hear echo its actually being caused by the other phone. check
your zapata.conf file and see if echotraining=yes.
Echo can be caused by lots of things so this is just a starting point.
For several months now we’ve been experiencing a really strange problem with
sound which best can be ex
I have the same setup and I don't have the problem you are having. The only difference I can see between my setup and yours is that instead of 'Playback', we are using 'Background'. Hope this helps.Thanks,Kyle
On 3/22/06, Sheeju .R.Alex <[EMAIL PROTECTED]> wrote:
Hi allI'm trying to dial out with
Is there anyway I can make one particular extension always dial out on
one specific pots line(group)
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Wai Wu wrote:
I don't know what the MIPS requirement is for trans-coding g729; but
only 4 Ts worth need to be trans-coded for my installation.
Then you will be near the max at 200% (100% for each CPU). I would
recommend a max of 3 T1s to keep the load under 75% for each CPU. It
might be
I am having a problem with asterisk not being stable enough for production use. I have two cards, the digium TE406P, and the TE410P. The TE410P is the primary card that I am using but I would like to move to the TE406P for the echo cancellation and more flexibility of PCI slots available.
General
I am intending to rebuild our asterisk server on a Dell 2800 with Dual CPUs.
I am currently using Centos release 3.5 (Final) with the 2.4.21-32.0.1.ELsmp
Kernel on a Dell 1750.
I have a TE410P (Quad PRI) card and a TDM card.
I remember that there were issues with the 2.6 Kernel, but Iam sure that
On Thursday 23 March 2006 02:08, Dr. Michael J. Chudobiak wrote:
> > I have hit a wall configuring a TDM400, I have set these up before
> > without issue but today I just can't seem to figure out what I am doing
> > wrong.
>
> I couldn't make TDM400/FXO work on my 1.2.5 Asterisk either. It wouldn't
We will do some test and if the load is too high, we will go with
extenal trans-coders boxes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Latham
Sent: Wednesday, March 22, 2006 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I don't know what the MIPS requirement is for trans-coding g729; but
only 4 Ts worth need to be trans-coded for my installation.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andres
Sent: Wednesday, March 22, 2006 3:56 PM
To: Asterisk Users Mailing Lis
Thanks Aaron
Worked a treat
Paul
- Original Message -
From: "Aaron Daniel" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, March 21, 2006 10:37 PM
Subject: Re: [Asterisk-Users] Cisco 7960 - Have to press a menu button to
dial
You h
mustardman29 wrote:
I might as well jump in. I am not clear on what the problem is but whether
it's a problem on something that needs to be done frequently or infrequently
or perhaps can be avoided with little effort, it's still a problem.
Your argument is more like the classic "it's not a bu
on half of the inbound (less than 48) to sip phones with g729, load
will be high, but that is a lot of licenses to buy so I will suspect
that you will end up using ulaw/alaw.
On 3/22/06, Andres <[EMAIL PROTECTED]> wrote:
>
>
> Wai Wu wrote:
>
> >What will my CPU utilization like? Will it be like
I was going to avoid naming names :P But anyway.. yes it's
asterlink. Guys seem nice enough.. and by golly.. when I switched to
SIP the termination is crystal clear... so far I'm happy with the
service from Asterlink... just wish I could use IAX2 oh well..
it really matters not to me HOW I
On Monday 20 March 2006 18:47, Matt wrote:
> "We have recently become aware of an issue in the chan_iax2
> implementation of IAX2. This issue leads to degraded audio quality.
> Due to this we are urging everyone to move to SIP."
That's from Asterlink; The way that they handle their IAX2 registrati
Time Bandit wrote:
If you have FOP, and if the call come in thru a ZAP channel, you can
drag the ZAP channel to your extension. This should work.
As for a way to make this happen from the manager API, I don't know.
Okay, thanks. We found a way to do it through the manager (i suppose FOP
does i
On Monday 20 March 2006 23:06, Ira wrote:
> I doubt the goal is to keep traffic down, in my case it might mean I
> don't need to delete all of the hardware related stuff which at this
> time is essentially of no interest to me. Personally I'd be more
> likely to help with answers to my limited ab
Wai Wu wrote:
What will my CPU utilization like? Will it be like 50% or more?
with g729 on 8 T1s, it will be like 400%.
Andres
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users ma
I've followed the post below and have just acquired a second-user Option 11c
system (rls 23.47 in the UK) now sitting on our testbench. I've tried all
combinations from various posts to get this to work with our Digium TE405P
but no luck. I suspect it's our PRI in the Option 11, it's an NTAK79
I might as well jump in. I am not clear on what the problem is but whether
it's a problem on something that needs to be done frequently or infrequently
or perhaps can be avoided with little effort, it's still a problem.
Your argument is more like the classic "it's not a bug, it's a feature".
What will my CPU utilization like? Will it be like 50% or more?
From: [EMAIL PROTECTED] on behalf of Andrew Latham
Sent: Wed 3/22/2006 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can this box handle 8 T1s (PSTN
Andrew Kohlsmith wrote:
On Wednesday 22 March 2006 10:51, John Novack wrote:
In fact, since the subject keeps coming up, perhaps there really is an
issue that needs to be addressed??
It has been addressed. There are a host of forum sites specific to Asterisk, and even more (such as
If you would be willing to make available for download the Adit 600 Install
/ Configuration manual for this unit I would gladly PayPal you for your time
and troubles...
TIA
Bart
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easyne
> the DIALSTRING you were given is just an extension, 089324154332. As
> Lenz pointed out, and it also says in the app_dial.c:773 WARNING, it
> must be technology/number, not just a number. Not sure perl methods,
> but you might concatenate a technology before the number, something
like
> $res =
> That's way too much Java for me. I'm lost already.
>
Doug,
I'm a Perl guy myself, so I think in terms of Perl and CPAN. I'm sure
Python has its own version of CPAN where people upload modules for other
programmers to use. CPAN has a Perl module:
POE::Component::Client::Asterisk::Manager
It'
Vegastream also will shortly have availability on the Vega 50 6x4, which
comes in a few different configurations including:
4FXS / 2FXO - MSRP $550
4FXO - MSRP - $600
8FXS/2FXO - MSRP $1000
8FXO - MSRP $1100
24FXS/2FXO - MSRP $2300
24FXO - MSRP $3000
48FXS/2FXO - MSRP $4000
These will support S
How about using CNG (comfort noise generation) on the client? That might do
the trick. Any better ideas?
>Asterisk does not support silence suppression - you should configure the
>cleints not to suppress silence. Not sure if "keep alive" packets would do
>the trick.
>Rob
>On 22/03/06, Steven Lan
Yup, even provantage has it listed as "Special Order" for $448
http://www.provantage.com/d-link-systems-dvg-3004s~7DLNH004.htm
On 3/22/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
> It has shipped, but availability has been sporadic.
>
> Cory Andrews
> Purchasing Manager
> ++
> VOI
D-Link has a 4 port FXO device on their site.
http://www.dlink.com/products/?sec=2&pid=451
Apparently it hasn't shipped yet and costs $500.00
I've been testing a AudioCodes MP104-FXO-C3S (around $1000) 4-port FXO
box. It works, but the number of configuration options are staggering,
complex,
yes
On 3/22/06, Wai Wu <[EMAIL PROTECTED]> wrote:
>
> Hi all,
>
> I am handed a project to setup *. The requirement is that it can handle 8
> T1s. Half of the calls coming into the system will be routed to SIP
> extensions (with transcoding). The machine we have in our disposal is a new
> dual Xe
Erik Anderson wrote:
> On 3/22/06, Andrew D Kirch <[EMAIL PROTECTED]> wrote:
>> Andrew D Kirch
>> Indianapolis, United States
>
>
> Well if that isn't one of the most bizarre emails I've seen come
> across this list.
>
>
> --
> Erik Anderson
> http://andersonfam.org
> __
On Mar 22, 2006, at 5:31 AM, Bjorn O wrote:
Hello all! For several months now we’ve been experiencing a really strange problem with sound which best can be explained as choppy/stuttery, and with a touch of echo on top. Basically, parts of a conversation might be choppy, but often combined with so
Tzafrir Cohen wrote:
On Wed, Mar 22, 2006 at 10:54:50AM -0500, Kristian Kielhofner wrote:
Hello everyone,
I am working on something now that could really use a snapshots.
For those that are not familiar, basically what it involves is having a
server with httpd running automatically checko
I am looking at using these on a overhead paging job to connect to
some paging adapters.
The costs are fine as long as it performs the function.
On 3/22/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
>
> On Mar 22, 2006, at 10:32 AM, Andrew Latham wrote:
>
> > I think you would need to alter the f
It has shipped, but availability has been sporadic.
Cory Andrews
Purchasing Manager
++
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Poor Andrew, everyone just comments how cool his email is ;)
I think the problem is:
exten => 2,3,Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
should be
exten => 2,3,Set(DB(forward/${CALLERIDNUM}) = ${FORWARD})
Note removal of the "$ {" and the "}"
good luck!
Andrew D Kirch wrote:
An
On Mar 22, 2006, at 10:32 AM, Andrew Latham wrote:
I think you would need to alter the firmware to set the kewlstart to
FXS instead of FXO. This is just a thought, I have not done such. I
decided that if it worked then such a device would have been marketed
already.
D-Link has a 4 port FXO dev
Hi
all,
I am handed a
project to setup *. The requirement is that it can handle 8 T1s. Half of the
calls coming into the system will be routed to SIP extensions (with
transcoding). The machine we have in our disposal is a new dual Xeon
3.2gHz server with 2g of ram and an dual 1000mb nic.
Austin Denyer wrote:
>Matt Roth wrote:
>
>
>>I think this is "The Last Starfighter" of Asterisk. If you solve this
>>problem in a timely manner, expect to be taken away by aliens to help
>>them develop their VOIP networks.
>>
>>
>
>I thought he was going to offer me "FIFTY TWO MILLION US DO
Matt Roth wrote:
> I think this is "The Last Starfighter" of Asterisk. If you solve this
> problem in a timely manner, expect to be taken away by aliens to help
> them develop their VOIP networks.
I thought he was going to offer me "FIFTY TWO MILLION US DOLLARS" for
assisting with an internationa
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