Can the asterisk system support video conferencing?
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I am planing on doing it a daemon that can live on the asterisk box or any
box that can run unix and iptables. I will need to reroute packets aimed for
providers box to the box where the daemon lives. In my case using a low
power(15watts) is the way to go. If your asterisk box has the spare power
Original Message
From: Eck [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Sent: Saturday, December 31, 2005 8:26 PM
Subject: RE: [Asterisk-Users] voicemail/privacy system
If you dont want to get too stuck into the guts of Asterisk yet, the
[EMAIL PROTECTED]
Original Message
From: Peter Bowyer [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, December 31, 2005 11:34 AM
Subject: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Hi all
Slightly OT but I know a lot of GS experts hang out here - I just
upgraded a GXP-2000 to
Well, the documentation states that Video Conferencing is possible. I've
tried working with EyeBeam, which yielded nice
Results, but anything beyond that - I can't comment.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Sunday,
Well, the documentation states that Video Conferencing is possible. I've
tried working with EyeBeam, which yielded nice Results, but anything beyond
that - I can't comment.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Sunday,
I am proposing an Asterisk system of many servers to service multiple
departments in a number of locatations to a large client. They have an
IT department but their Linux skills are weak and they are likely to
face a high churn rate in staff so it would not be wise to expect a high
level of
I saw a brief discussion via Google about
developing support for LCD displays, ones that you integrate into a drive bay or
whatnot for server information output. Any development on this? I
couldn't find much.
--Mike
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On Sat, 31 Dec 2005 20:23:09 +0100, BJ Weschke [EMAIL PROTECTED] wrote:
From the Asterisk CLI you can do show queues and show agents.
There are also a number of third party tools, free and not-free, to
take information from Asterisk and present it in real-time and on a
historical basis.
[EMAIL PROTECTED] wrote:
I am proposing an Asterisk system of many servers to service multiple
departments in a number of locatations to a large client. They have an
IT department but their Linux skills are weak and they are likely to
face a high churn rate in staff so it would not be wise to
I also had DTMF problems with my TDM400 when I upgraded from Asterisk 1.0.9
to 1.2.1. After the upgrade I noticed that my bank IVR and work VM would
not recognize DTMF coming from my * system. I had to add the 'toneduration'
parameter and bump it up to 300ms before it began to work correctly
you want something really cheap. you got to visit www.broad-tel.com. It is even offering a WiFi phone at US$125 for its existing clients.
On 12/20/05, Dakota [EMAIL PROTECTED] wrote:
Are there any IP Phones that can work with Asterisk, that cost less than $60?if so, what's the model/manufacturer?
Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401
response from Asterisk 1.2.1 server. A few minutes later is
registered again.
It happend at least two times since Asterisk version 1.2.1 is used
at the server, but I am not shure if the problem already existed
before this update.
Has
There will be one announced at CES next week by a major
company.
Kerry
GarrisonPublisher - http://GeekGazette.com - http://VOIPSpeak.net
(949)502-7819 x200- [EMAIL PROTECTED]http://www.techdatapros.com
On 12/20/05, Dakota
[EMAIL PROTECTED]
wrote:
Are
there any IP
There are4 options for your consideration:
1. use 2 x 1-port FXO gateway
2. use 2-port FXS gateway with FXS to FXO converter
3. use a 4-port FXO gateway.
4. use 2 x X100P cards
You can get them from www.broad-tel.com
On 12/21/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm looking for a
Hi James,
This link might help:
https://billing.atlasvoice.com/forum/index.php?topic=20.0
-- Bjorn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Ronald
Sent: Sunday, January 01, 2006 1:56 AM
To: Asterisk-Users@lists.digium.com
Subject:
On Wed, Dec 28, 2005 at 04:02:00PM -0800, William Boehlke wrote:
The 830s are nice but limited because they do RAID on a card and have but
one suitable PCI slot. So you can have an interface card or RAID, but not
both.
Linux software raid is, in our experience, much better than any hardware
What does this warning mean?
WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on
REGISTER that isn't a register
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On 1/1/06, Robert La Ferla [EMAIL PROTECTED] wrote:
What does this warning mean?
WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on
REGISTER that isn't a register
Your SIP device is returning a 200 OK message about a registration
attempt, but Asterisk doesn't believe
Leif Neland [EMAIL PROTECTED] writes:
My GS BT101 have also developed problems with sync'ing to my ntp-server.
I can see, using tcpdump, that the phone asks my server and gets an
answer, but the display is not updated.
It used to work, but now it usually doesn't, but strangely, sometime
it
Bjorn,
Thanks!! The example looks like what I need although I don't understand the
Forward to VoIP as I don't have an ITSP. I'll give it a try later today.
Basic Settings:
Number of rings: 1;0 is not a valid option
Forward to VoIP: a number in your from-pstn context where you want
Hello Kerry,
Maybe it's me, but why are you using hint in this fashion? Shouldn't
you be doing exten = 100,1,Dial(SIP/900zap/g0/w5551212) or is there
something new that I have missed?
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 01, 2006 11:42 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Having major issues with TDM2400
Hello Kerry,
Maybe it's me, but why
Moises Silva wrote:
Yep, perfectly possible. I would do that with AGI and php, in your case,
perl works as well.
The only thing you need is read documentation regarding AGI, Voicemail
and
extensions. Its kind of difficult to helo you further if you dont tell
us
how much you know about
On 17:29, Sun 01 Jan 06, Stefan Tichy wrote:
Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401
response from Asterisk 1.2.1 server. A few minutes later is
registered again.
It happend at least two times since Asterisk version 1.2.1 is used
at the server, but I am not shure if
On 21:40, Sun 01 Jan 06, Michiel van Baak wrote:
On 17:29, Sun 01 Jan 06, Stefan Tichy wrote:
Snom 190 phone (snom190-SIP 3.60k) occasionally gets SIP 401
response from Asterisk 1.2.1 server. A few minutes later is
registered again.
It happend at least two times since Asterisk
Cell Socket is another such product.
Current Cell Sockets work with some of Motorola phones.
Different systems GSM, CDMA, work somewhat differently regarding
callerID and speed dial
The original CellSocket worked with certain Nokia phones
In the GSM version dialing is similar to the PSTN, but
Yes,
I got the same error when I tried to register my G.729 license.
When you downloaded the patch, are you sure you did that on binary or ascii?
My problem was my download was automatic. I forced to binary and the problem
was fixed. Check the size of the files, on your machine and the ftp
On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote:
To summarize, I spent 6 hours yesterday on the phone with Digium trying to
fix a problem with the TDM2400 ad we still don't have it working right. The
lastest version of everything are installed and confirmed by Digium. So here
is the issue:
On 1/1/06, VoIP Newbie [EMAIL PROTECTED] wrote:
There are 4 options for your consideration:
1. use 2 x 1-port FXO gateway
2. use 2-port FXS gateway with FXS to FXO converter
What is an FXS to FXO converter? you have any URLs?
3. use a 4-port FXO gateway.
4. use 2 x X100P cards
You can
The goal is to create a user that has a SIP device and a custom ZAP channel
device, have them both ring until one is answered, basically a ring group.
But I am using AMP's users and device mode rather than the extensions mode.
I have this working properly on my office system. However, with the
Waldo Rubinstein wrote:
I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic
PBX to Asterisk. Can anyone recommend a stable and reliable one?
Use 2x Sipura SPA-3000 - and you will also get 2x FXS...
Or use a Digium TDM02B (2x FXO).
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
The goal is to create a user that has a SIP device and a custom ZAP channel
device, have them both ring until one is answered, basically a ring group.
But I am using AMP's users and device mode rather than the extensions mode.
I have this
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
The goal is to create a user that has a SIP device and a custom ZAP channel
device, have them both ring until one is answered, basically a ring group.
But I am using AMP's users and device mode rather than the extensions mode.
I have this
Hello Kerry, I do it exactly as such, however in steps. My
understanding of the hint system is just for notification of status, not
for execution of dialing.
I regularly use this same setup you are looking for, rings in, then
rings 2-5 devices (some zap, some iax) and the first one that answers
Oh just a followup, if you are trying to do an outbound dialout over
analog, what others are saying is correct. You could consider however
using a voip provider to make the outbound call, then you should have
status.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Thanks everyone, the reason I posted here was because a Digium support tech
said it should work and he couldn't figure it out. So while I appreciate
everyone's comments that it wont work, a technician from Digium said it
should, hence I turned to the list for clarification. This is not really a
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Thanks everyone, the reason I posted here was because a Digium support tech
said it should work and he couldn't figure it out. So while I appreciate
everyone's comments that it wont work, a technician from Digium said it
should, hence I turned
I did not say I had a problem with support. The problem was the tech ran out
of time on Friday and there was nobody to escalate the problem to. So
instead of waiting until tomorrow for teir 2 support, I looked to the people
on the list to see if I could find an answer before then. It seems as
On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
Thanks everyone, the reason I posted here was because a Digium support tech
said it should work and he couldn't figure it out. So while I appreciate
everyone's comments that it wont work, a technician from Digium said it
should, hence I turned
Are you using raid for performance or redundancy? Software raid is nice
except when the drive that fails is the one with your boot partition on it.
I guess you could always tftp boot the kernel or something.
Craig
- Original Message -
From: Louis-David Mitterrand [EMAIL PROTECTED]
As much as I like the option of implementing a follow-me type of script, the
second problem is that the client wants to use AMP to manage the extensions.
Just doesn't seem like I have a solution that fits all of the client's
requirements. The easiest solution seems to be to use a SIP trunk for the
Perhaps a Sipura-3000 could be of use here? Anyone have any ideas about
that?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 10:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Well, it would have to be 4 of them for each of the available PSTN lines. I
have also considered a Mediatrix channel bank.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 01, 2006 7:53 PM
To:
[EMAIL PROTECTED] wrote:
Well yeah, I had no intention of buying one, I was just wondering what
the hell it actually was that the seller was trying to hide.
Their supplier?
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Asterisk-Users
www.antek.com.tw
Had 4 port fxo, for around 200 to 250$
They are OEM, and can change things if u need.
I tested it breifly in there office last year in Computex 2005
You can contact [EMAIL PROTECTED] for wholesale.
Rehan
On Fri, 2005-12-30 at 17:53 -0800, [EMAIL PROTECTED] wrote:
Hiu Yen Onn wrote:
How big of RAM for Asterisk server? My production environment will be
about 400 users in the office.
In one server? 4GB. And more if you can.
I'd suggest you use several servers for 400 users unless the percentage
of active phones is ~10%.
Mike
Chris Mason (Lists) wrote:
I am proposing an Asterisk system of many servers to service multiple
departments in a number of locatations to a large client. They have an
IT department but their Linux skills are weak and they are likely to
face a high churn rate in staff so it would not be wise
Hi,
Not very reliable for commercial setups, they do have issues hanging up
ports etc. Quintum over Antek any day.
Regards,
Sahil Gupta
VoiceValley
On Mon, 2 Jan 2006, Rehan AllahWala wrote:
www.antek.com.tw
Had 4 port fxo, for around 200 to 250$
They are OEM, and can change things if
Does Quintum has a 4 port fxo box ?
Hi,
Not very reliable for commercial setups, they do have issues hanging
up ports etc. Quintum over Antek any day.
Regards,
Sahil Gupta
VoiceValley
On Mon, 2 Jan 2006, Rehan AllahWala wrote:
www.antek.com.tw
Had 4 port fxo, for around
Hi I am trying to use g.726 so as to make calls, further i am using
cisco PAP ATA's, on these PAP's i have a number of options ranging
from 16 to 64 kbps for g.726, i wud prefer to use the 16 kbps version,
as in it is my sip.conf i have done this
allow=g726
On the PAPS i have selected g.726-32
5)
How do I change the time zone for Asterisk? Currently the system time is
correct but when I dial *60 it reports a different time (out by many hours).
I'm not familiar with this option. Can you please tell me more or send
me some link.
FYI, this is the relevant
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