I'm curious, how are you licensing your codec? The source is open, but the
codec usage licensing is not. I think you'll find that licensing it from
Digium will be much simpler, not to mention their code will Just Work(tm)
without any messing around.
-d
At 12:08 PM 4/17/2005, you wrote:
Hi,
I
Hi all,
I've put together a quick petition, in hopes that we can possibly persuade
Sipura (or any other large-scale IP handset manufacturer) to include
firmware support for IAX. The IAXy has proven that an IAX product is in
demand, and very useful, and I think we'd all like to see a handset
First, I suppose, you'd have to write it ..
AFAIK, the webmin project was abandon a couple of years ago. I don't think
it was ever even remotely near being completed.
-d
At 05:25 PM 3/31/2005, you wrote:
How do I install the asterisk module for webmin?
Look at bkw's valet parking
-d
At 03:58 PM 3/21/2005, you wrote:
I'm working through my list of features people will expect, and Hold
Pickup is at the top at the moment -- has anyone done any work on
this? We've had some unpleasant experiences with call parking, and
everyone seems to like the
Isn't the idea of this, to sort of use the desk as it's plane to pick up
sound? I thought a few vendors did this kind of thing...
-d
At 03:33 PM 3/7/2005, you wrote:
The microphone is [somewhat inexplicably] mounted in the base over a hole
that
faces downwards, between two of the rubber feet,
I don't know - they look kinda lame. I mean, why is their SIP server
seemingly better-routed than their IAX server? In my case, their IAX server
is almost 20ms further away than the SIP one -- seems odd to me.
Think I'll stick with Nufone - very well routed, and only ~15ms away. :)
-d
At 06:37
Why would you even want SSH exposed to the world? In fact, why expose it to
anything but your local admin console, or *maybe* a vpn tunnel server if
absolutely necessary?
-d
At 10:08 AM 2/10/2005, you wrote:
The hack came in through ssh.
IMO, your best defence is an extremely strong root
Hi Tim,
No hardware - been done, see rxfax. Dumps it to a tiff, you can do
whatever you want with it (email it out, convert to pdf, send to a
printer .. OCR and voice to speech it and play it over the PA system ...
:)
-d
At 01:07 PM 1/31/2005, you wrote:
How
hard would it be to write (or has
?? u only so whats going to determine its a fax ? you
need a dedicated box if you have no line card in the box. asterisk + SIP
is not capable of determining whats a fax and whats voice. you need
a card unless you have a dedicated number
On Mon, 2005-01-31 at 13:12 -0600, denon wrote:
Hi Tim
At 01:49 PM 1/19/2005, you wrote:
There are systems that use G.711 when traffic is light, but
switch to compression codecs under heavy traffic to conserve
bandwidth. I don't know how/if this can be done in Asterisk.
--Stewart
I don't think there's anything like that built into * as it is now,
So? it's a big list .. I'm sure if you'd like to donate some quad xeons and
gigE pipes, it could be resolved very quickly..
-d
At 04:41 PM 1/13/2005, you wrote:
OMG! 1 hour?!?! I just now got this at 4:40PM. It takes an hour for my
emails to get posted to the list? Geez..
-Matthew
- Original
That's data or fax CNG, not dtmf. And yes, it's disabled for the duration
of the fax or data session.
-d
At 11:36 PM 1/10/2005, you wrote:
Hello all,
I am getting console debug messages about tone detected on channel XX,
disabling echo cancelation on channel XX when using echocancel=yes with a
Old news, Asterisk 1.0 released .. :)
Here's another mirror -- should be very fast from most anywhere. Take it
easy on Digium's bandwidth. :)
http://asterisk.paperwork.com
-d
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
You can snag em from http://asterisk.paperwork.com and if you drop me a
note with your url, I'll add it to the list.
-d
At 10:11 AM 9/23/2004, you wrote:
If anyone who got the 1.0 tar's would be able to get them to me I'd be
more than willing to donate traffic toward the effort by mirroring it
hehe .. I think we have more bandwidth than sourceforge now.. I've got like
9 on my list now.
-d
At 10:58 AM 9/23/2004, you wrote:
Maybe someone should make a bittorrent? I will contribute some BW
if there is a torrent.
Steve
Kenneth Shaw wrote:
To be Slashdotted within 30 minutes.
-Ken Shaw...
Keep in mind, PPTP will only tunnel through the NAT, as long as GRE (prot
47) is properly tunneled along with tcp 1723. This support is relatively
standard in common NATs, but it's not a given.
-denon
At 09:23 PM 9/21/2004, you wrote:
First,
I assume that you will be running NAT at both
I agree entirely! I've been told it's not possible, but I'd love for
someone to prove me/them wrong.
-d
At 11:54 PM 8/10/2004, you wrote:
Hi,
I was wondering if any CISCO users out there knows if it is possible to
Change the locations of the BUTTONS along the bottom of the screen.
I ask this as
Sure, head to :
http://store.yahoo.com/asteriskpbx/wildcardx100p.html
-d
At 12:25 PM 12/30/2003, you wrote:
I'm trying to buy a new X100P but
http://shop.store.yahoo.com/bsdmall/wisifxoin.html
is failing to check the order
Anybody knows any other way to purchase it?
Isamar
I've been having issues getting FWD to work. I posted this same Q to the
FWD forum (no responses yet), but I was hoping someone here had some insight:
http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0
I just signed up for an FWD account (I know I
Hi all,
I know there have been some neat changes to how call recording works
lately, and so therefore it's much easier to do .. but I haven't been able
to find any details in the archives about the best way to accomplish it
these days. Does anyone have a sample of their config they'd share,
That works better here in Minnesota .. Land of 10,000 Lakes .. go pick one
and jump. :)
-d
At 05:16 PM 12/12/2003, you wrote:
Instead of quacking out useless information, it's more useful to not
answer.
Their are alot high places on this planet.. pick one and jump.
bkw
Depends if you're phone supports it, and you have reinvites etc enables
in *.
-d
At 03:17 PM 12/8/2003, you wrote:
Hi
all,
Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the
two sip phones( hard or soft)
Some ADSI phones come locked to a certain service provider. You cannot
load your own adsi scripts into these phones - you need one that isn't tied
to a specific company or pbx.
-d
At 06:35 PM 8/27/2003 +0200, you wrote:
Hi,
one question:
What you mean with unlocked ?
-Ursprungliche
Adtran 750 channel bank and a T100P.
-d
At 01:06 PM 8/19/2003 -0300, you wrote:
Hello,
I am looking for hardware for
Asterisk.
I want to connect analog lines (from 6 to 12 or
more) to Asterisk, what will be the best hardware for that?
Thanks,
Bartosz
Are these locked to the service, though? Look what vonage managed .. :)
-d
At 12:36 PM 8/17/2003 -0400, you wrote:
$75 for the single ethernet port version and $85 for the dual ethernet
port version.
You can get two for $129 at www.sipphone.com
-Original Message-
From: [EMAIL
Last I checked, SIP transfer to park doesn't work .. only way to do it is
using T and a # transfer .. which is ugly. Has this been fixed?
-d
At 10:51 AM 7/30/2003 +0200, you wrote:
park the call
On Wednesday 30 July 2003 11:16 am, [EMAIL PROTECTED] wrote:
hi,
can someone who has used
Yes, you can do all this ..
an X100P and two TDM400s, one with all 4 FXS, one with only 1 FXS.
-d
At 07:53 PM 7/12/2003 -0400, you wrote:
Hi all,
I need a phone system that has 2 incomming lines and 5 extentions.
I will need music on hold
I need all incomming callers to get a message: press 1
I had this a while back, and set canreinvite=no, and it fixed it.
-d
At 08:42 PM 7/11/2003 -0700, you wrote:
I've been banging my head on this for several hours, and I have no idea
what's going on. Maybe there is a very simple result, and I've been
looking too hard at this this evening.
a callerid string..
-denon
Thanks,
Steve
___
Steve Creel[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
What'd this device set ya back? Have a url?
-d
At 11:45 PM 6/30/2003 -0700, you wrote:
AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with
SIP up and running with *
Michael Kane wrote:
The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8
I just picked up a couple CAC Access Bank 1s loaded with FXS that should be
arriving shortly. Does anyone have one that they use with Asterisk? If
so, would you be willing to shoot me a note with your current configs? I'm
not very familiar with CAC/etc, and it would save me countless hours of
Maybe the hardware date?
-d
At 05:07 PM 6/17/2003 -0400, you wrote:
Where is this time that is sent to the phones with the callerid info
coming from ?
If I do date at the command line I get the correct time as set by ntp
yet the time the phones get set to is 50 minutes slow.
Hi all,
I've put together a quick Asterisk survey, in an effort to attempt to get
some insight into the overall direction of the project. I'd appreciate if
you could spend a couple minutes and run through this - I'll be happy to
share the detailed results with Digium or anyone else who's
We're doing a new * installation at a remote office soon, and I was just
curious what people's opinions were on hardware these days .. I've had
decent luck with T100Ps and Adtran, but I know times change ..
I'm looking to do roughly 15 handsets and 15 pstn, with some room to
grow. I had
Hi all,
I've got a fairly minor question, but it's getting on my nerves ..
hopefully it's an easy answer. I'm having trouble parking calls on our
7960s. It works fine on ZAP devices, though, and they're both using the
same context.
What I do is:
When I'm on the call, I hit More, Transfer,
At 03:31 PM 5/31/2003 +1000, you wrote:
I've seen exactly two people with this problem and it seems to be related
to the system that it's running in, although I don't know if it's power
supply, motherboard, or peripheral related, but somehow noise on the
system power supply creeps into the
I've got an end-user in Australia that needs some gear .. I can ship it
over there, but it'd be a bit of a pain to find adapters localized to AU,
even though I know stuff is all 110/220 these days.
Anyone know a good cheap place to buy a Cisco ATA186 and maybe a
Netgear/etc dsl router/switch
2003, denon wrote:
Does your link button work, after you program it with adsiprog? It broke on
my 350 .. had to clear the asterisk load out again to use it. Link/flash
is sorta important to an asterisk phone...
At 08:21 PM 3/27/2003 -0500, you wrote:
Well, I almost cried... Here's why:
I
I'm having some problems getting an ATA186 behind NAT working. When I had
it on the same subnet as the Asterisk server, it worked fine. Now Ive
taken the ATA on the road with me, and it's behind a Dlink router+firewall,
doing NAT. I pick it up, hear a dialtone .. the firewall on the asterisk
Oh, and yes, the * is current as of a few days ago .. so it should have
that new SIP code mark was working on a while back.
Thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
in your dump it
says non-NAT
Mark
On Fri, 21 Mar 2003, denon wrote:
Oh, and yes, the * is current as of a few days ago .. so it should have
that new SIP code mark was working on a while back.
Thanks
___
Asterisk-Users mailing list
[EMAIL
already working to implement IAX support, so you'll need something unique -
whether it be price or unlimited international, etc.
denon
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
utilization.
denon
At 09:19 AM 3/13/2003 -0600, you wrote:
What do you all think of renaming IAX2 as:
Telephony Authentication, Signalling, and Transport Exchange (TASTE)
TASTE is easy to remember and has a sort of ironic relation to SIP.
Is it took hoaky?
Mark
Nope, that would require simulating the a fax dsp, or such. You can,
however, route a call to a certain zap port when a CNG tone is detected.
At 01:36 PM 3/3/2003 -0500, you wrote:
Thanks Martin, I'll set it up tonight. Can asterisk receive the incomming
fax and store it to local disk with
44 matches
Mail list logo
