Hi Curt,
I saw an example of this at Astricon last
week. The guys were from Sweden
and they installed it onto every Asterisk their company installed as part of
the base package.
An end user could log in from within the
company lan or externally to perform the following functions;
2006/10/31, Jon Farmer [EMAIL PROTECTED]:
Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information:
*/SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP
MW == Mike Williams [EMAIL PROTECTED] writes:
MW The control connection (port 5060) obviously goes via the asterisk
MW server as it has to work out where to send the control to, but I
MW could quite easily imagine the audio going directly handset to
MW remote server or handset to asterisk to
Hi everybody,
I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work withnew versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should
Ahh yes so it is. Thanks for the pointer.. seems fairly straight
forward for an upgrade.. guess a test systme is the only way to know
for sure :)
On 11/1/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Sorry, the file is located here:
[EMAIL PROTECTED] ~]# ls -l
Hello,
I had had some echo issues. I purchased a digium echo canceling card,
and the echo issue seems to be reduced but not eliminated as I hoped
it would be. I currently have it set to 128 'yes'. I've tried 256,
but when I try 256 what happens is that instead of getting better echo
canceling
Doug,
I've seen that and a similar thing with reversed from and to fields on a
reinvite issue I'm having in 1.4. Olle was going to look at it.
Kevin Collins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, November
Do the speakerphones work well on
Snom 320s? I have a Linksys 841 and could never get the
speakerphone working well.
-CP
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes
Sent: Wednesday, November 01, 2006
11:45 AM
To: Asterisk Users Mailing List
After doing some research on the Internet and studying all the major IP phones, I have came to a conclusion that Grandstream GXP-2000 has the most features of all the phones for the least price of all. I don't know how they are managing to manufacture their product for such a cheap price, but
Hi Vladimir,
If you mean the Mexuar Corraleta
application that was launched at Astricon last Tuesday by www.Mexuar.com it is available here;
http://www.mexuar.com/products_sdk.shtml
There is a press release for you to read
here;
I came to the same conclusion.
There is one thing however that the GXP2000 needs in my opinion.
There is no dial plan avaiable in the configuration, this means that when
dialing a number there is a slight delay before it actually dials.
With a dial plan the dialed number is sent immeadiately the
It seems that when you use Realtime
static and possibly realtime realtime for sip users, that Asterisk fails to
create the regexten context for DUNDi.
Someone else had the same problem back in July. Doesn't
look like they ever had a resolution.
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 01, 2006 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote-Party-Id and Attended Transfers
Douglas Garstang wrote:
Has
Hello all,
This is my first message to the mailing list. I am seeking advice as to
how to proceed/what to get for my current situation.
I want to use asterisk to make a system that does pure outbound calls
and plays a message upon a live answer or answering machine. Basically
it needs to
Can someone get this guy off the lists?-- Forwarded message --From: [EMAIL PROTECTED]
[EMAIL PROTECTED]Date: Nov 1, 2006 3:22 PMSubject: Benachrichtung zum =?unicode-1-1-utf-7?Q?+ANw-bermittlungsstatus (Fehlgeschlagen)?=To:
[EMAIL PROTECTED]Dies ist eine automatisch erstellte
Hello,LookX-web litehttp://www.asterisk-es.org/modules/mydownloads/visit.php?cid=6lid=12Regards
On 11/1/06, Vladimir Montealegre Estailes [EMAIL PROTECTED] wrote:
Hello list partners
you know about a softphone made in java attachable
in a web page?
GNU!
Thaks in
advance!
Visita
Anyone know the cost?
Best regards,
Al Bochter
Bochter Services
(Voip PBX) Free World DialUp: 780217 EXT: 250
WebSite: http://www.freeworlddialup.com/
http://www.BochterServices.com/?t=Email
BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email
For new and used
Hi All,
I'm having problem with IAX, I'm trying to connect to speex.co.il
from asterisk using:
register = username:[EMAIL PROTECTED]
and I cant get it to work.
Maybe someone who already got this to work will help
When dialing my speex extension I see the next output from
I have two Sipura 3000s, one for our main phone line the other for our
fax line. I think I need to handle each device in seperate context
sections. Both contexts use the s extension and things are not working
as it was before I added the second Sipura for the fax line and
additional context.
We used the SPA-94x fro desktop phones and
the speaker phones on them a pretty good. We have a Snom 360 and the speaker
phone is lousy. I have just updated the firmware to the latest version and it
seems to be a better. It is not as good as the SPA as the Snom has background hiss
on
Stephen Bosch wrote:
Dovid B wrote:
Read the book Asterisk: The future of Telephony
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
It will teach you a lot.
The trouble with this (I have it) is that it's dated.
I do wish we had a more structured and maintained
The way you want to set this up it seem the best way to go about it,
is to have the Panasonic see Asterisk as a VM System, and have the
asterisk extensions connected to the panasonic using Analog ports on
the Panasonic.
What panasonic system is this?
On 11/1/06, [EMAIL PROTECTED] [EMAIL
We have just finished some Asterisk installs on Dell 2950's, and they
work flawlessly.
Has anyone done an install on an 860 as of yet?
Kind regards,
PaulH
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To
In addition, I have created a possibly larger dump of
the issue, as below. Can someone help me determine what the problem
is? Is there more information that I can provide? I am running libpri
1.2.3, zaptel 1.2.9.1 and asterisk 1.2.12.1:I´m using static realtime asterisk configuration
What does zap show channels show? Are all the channels shown as in
use? What is set in zapata.conf for resetinterval= ? If anything. I
believe that resetinterval is used to reset unused channels for any
channels that are left open.
On 10/31/06, Asterisk [EMAIL PROTECTED] wrote:
Hi All,
I have had the opportunity to test many IP phones in the last 6 months
and I thought you might enjoy a quick review of what I have found.
Grandstream Budgtone 200 - Poor Quality for business use- Looks good,
and the handset feels nice, buttons have a decent feel, but the disply
is
For Xweblite (which I dont think
does what the original person was after) no idea.
For Mexuar Corraleta Technology SDK
$US2,000 per server, unlimited throughput
http://www.voip-info.org/wiki/view/Mexuar
Regards,
Dean
Collins
[EMAIL PROTECTED]
+1-212-203-4357 Ph
I think this page will get you on the right track:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
bp
On 11/1/06, Zak Kinion [EMAIL PROTECTED] wrote:
Hello all,This is my first message to the mailing list.I am seeking advice as tohow to proceed/what to get for my current
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 01, 2006 9:57 PM
Subject: Re: [asterisk-users] simultaneous ring - call groups or
queuesorsomething else?
Hello Matthew,
Did you test Snom or Sipura hard ip phones? I was considering Budgetone for an
office of 10 users. After reading your testimonial I will have to re-think my
selection.
I do wish to continue having dialogue about this very issue.
Ed
Sent from my BlackBerry® wireless handheld
More an echo algorithm question than a purely asterisk one...
I have the following setup:
Handset - PAP2 - Asterisk - SPA3000 - Telco
And no matter what I do, I get echo on a call routed out via the PSTN
when I talk into the handset, in the order of a hundred ms (my estimate,
could be wildly
I remember when I started working with the manager (I was using VB) that you
have to send a string of characters that you are going to the next line. I
am not sure if this will solve the issue. (sorry if my response dosent make
much sense - a bit on the tired side)
- Original Message
On Wed, 1 Nov 2006, Matthew Mackes (Webmail) wrote:
Asterisk will be running on Pound Key Linux, on three HP Servers- All
DualCore Xeons, Dual Processor machines, (so four cores per machine) with 4
GB of RAM per. We will also be connecting the machines with Gbit Ethernet to
one another on a
I can't even get regexten to work with config filesOn 11/1/06, Douglas Garstang [EMAIL PROTECTED]
wrote:
It seems that when you use Realtime
static and possibly realtime realtime for sip users, that Asterisk fails to
create the regexten context for DUNDi.
Someone else had the same
Thank you for your message-
I have held a Linksys/ Sipura. It looks nice, however the model I saw
was $125 for two line appearances. Aastra offers 12 line appearances in
their 9000 series phone for about the same price. Check out
www*aastratelecom*com/ipphones/pro_240*asp
I have found that
Wow--- Good to knowThanks!!! But does that mean that you are not
using your other processors?
How many calls are active on one server at one time?
Steve Edwards wrote:
On Wed, 1 Nov 2006, Matthew Mackes (Webmail) wrote:
Asterisk will be running on Pound Key Linux, on three HP Servers-
I've got CentOS 4.4, Asterisk 1.2.9 (TxBx 1.1.1) on four Dell PE2800's
with dual P4 3.4Ghz CPUs and 2GB of memory each. These boxes run user
extensions for seven of our offices (3 in the USA, 2 in Europe, 2 in
Asia), all of the meetme conferences - ~10/simultaneous/hr w/ approx. 10
people per
Does anyone have any experience with this? We're looking to deploy a
pretty robust HiDef Video Conferencing solution, and if it were built
around Asterisk, that'd be a huge bonus. It looks like a bounty was
offered on it for a while with no results, and now an Indian company -
Adiance - claims
Hi there
I also get this error:
Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
whenever I try to call my Alcatel 4400 PBX etxn from SIP using TE110P. The
output of zap show channels is:
Chan Extension
Hi,
Any one using Rubi asterisk manager interface
http://rubyforge.org/projects/rami/ ?
How stable/usable it is?
raj
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On 02/11/06, Matthew Mackes (Webmail) [EMAIL PROTECTED] wrote:
As far as Snow- They look very cool, and I love almost everything Linux
based- PDA's, PVR,s, everything- but, I wonder if it will need to be
rebooted every once in a while to stay happy- Every phone that is SIP
has an OS- so, its
Your right- I should- I have heard good things.
Stephen Davies wrote:
On 02/11/06, Matthew Mackes (Webmail)
[EMAIL PROTECTED] wrote:
As far as Snow- They look very cool, and I love almost everything Linux
based- PDA's, PVR,s, everything- but, I wonder if it will need to be
rebooted every once
Marlin Unruh wrote:
I have two Sipura 3000s, one for our main phone line the other for our
fax line. I think I need to handle each device in seperate context
sections. Both contexts use the s extension and things are not working
as it was before I added the second Sipura for the fax line and
Is it possible to connect two or more pbx systems over Ip via Asterisk?
The asterisk boxes should have FXO ports to connect to the local pbx
system right?
regards,
Christopher
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On Thu, 2 Nov 2006, Matthew Mackes (Webmail) wrote:
Wow--- Good to knowThanks!!! But does that mean that you are not using
your other processors?
How many calls are active on one server at one time?
Correct. cat /proc/cpuinfo shows a single processor. Someday I hope the
meetme/SMP bug
On Wed, 1 Nov 2006, Maxx Lobo wrote:
All the boxes have SMP kernels, and I've never had an issue with Meetme or
any other feature.
I wish :)
Maybe it's something funky with the HP implementation, but I've gone from
crashing with embarassing frequency to not a single crash with the non-SMP
Can someone please help me with a problem that I seem to
have with this Polycom 601 phone. It will not see my TFTP server and
keeps saying Could not contact boot server, using existing
configuration. I have Linksys phones that use the TFTP server
without any problems but this Polycom will
Henry.L.Coleman wrote:
Its a bit like the VHS vs Beta war, both systems have their good and bad
points In the end, sales/marketing perception will always win regardless
of better technologies.
That will be Skype then ;-)
--
Jon Farmer
Telford, Shropshire, UK
Ejay Hire wrote:
This is incorrect. The data is still packetized and passed through IP
are you sure? ;)
we can connect two zaptel channels directly (example - call from channel
bank to pstn. both connections to channel bank and pstn are e1).
___
Hi,
Is there any possibility to have md5 encoded passwords in the IAX users
database? I notice the secret AND/OR md5secret columns always have to
contain the password in plain text even when you set the auth column value
to md5?!?
Am I missing out something? Any ideas on how to correct this?
On 11/1/06, Rajkumar S [EMAIL PROTECTED] wrote:
Hi,
Any one using Rubi asterisk manager interface
http://rubyforge.org/projects/rami/ ?
How stable/usable it is?
It probably hasn't seen much use. I created that back when I was just
learning ruby, so it could probably use some refactoring as
Hi there,
It seems mpg123 is now maintained again and there are a few new
version released. The latest is 0.61. Does asterisk work with the
newer versions? What audio options shall we use when compile mpg123
v0.61?
Thanks!
Frank
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