from their site:
What technology does voiceglo use?
voiceglo uses a standard voice-over-IP protocol called SIP with
patent-pending software that allows voiceglo endpoints to work on IP
networks that employ address translations (NAT) and firewalls. voiceglo also
uses advanced voice compression
Philipp von Klitzing [EMAIL PROTECTED] said:
During first visit to Amsterdam by car (with a German number plate - aah
aah) exactly that cobblestone landed in my side window. Nothing stolen,
just a friendly welcome message... ;-
Heh, I recall the first time I went back to the Netherlands with
Hi
I want to connect a Windows machine over TAPI with the Asterisk PBX.
So is it possible to connect the Windows machine directly to Asterisk
(Zaptel card)?
Thanks
Harry Baron
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Dear all,
I'd like to know if the core (demo, voicemail...) asterisk prompts
have ever been recorded in french (and are freely available).
If not, I'm willing to have them studio-recorded by a professional
speaker, and contributed back to the community. Does a message list
exist ?
--
Nicolas
Cees de Groot wrote:
Philipp von Klitzing [EMAIL PROTECTED] said:
During first visit to Amsterdam by car (with a German number plate - aah
aah) exactly that cobblestone landed in my side window. Nothing stolen,
just a friendly welcome message... ;-
Heh, I recall the first time I went
I'm a windows developer ready to develop TSP/MSP (TAPI) drivers for asterisk PBX
but i searched a little more support to develop these drivers ...
unfortunatly i have to develop the drivers commercially because i will need to hire a
asterisk freak to explain me in detail how everything
Hi All,
I'm just testing * with MSN 4.7 and works great, but when i try to call using the dialpad of MSN
all my number into * appers twice.
Like 100 when i try appears 1 ..
Like a echo in the number.
I'm using rfc2833 because inband just crash ...
sip.conf.
[msn]
type=friend
host=dynamic
Florian,
Sorry you haven't heard anything but we've recently decided not to offer this product
out side of Holland. If your still interested we have another product called ISDN-Flex
that provides SIP/H.323 PSTN access inbound/outbound but you need to be connected on
on one of our IP or
Hi Lenny,
Thanks for your response. Does the *24 work for you, does it actually beep
?. The funny thing is, I came accross the A(x) option at www.voip-info.org,
but it's not being mentioned in the app_dial.c of Asterisk 0.5.0. So all
this left me in doubt about whether it actually exists. Since I
Hi
Here is my scenario
Mr.X's Asterisk Box Dials Mr Y's Asterisk Box (thru Zaptel channels)after
Channel establishment Mr. X send DTMf tones to Mr Y using by using
application SendDTMF().
My question is this is there any method that Mr. Y Saves these DTMF Tones in
any variable (after
Hi,
Can anybody explain me in configuring Asterisk to record a conference?
Regards...
Girish
_
Add zing to Hotmail. Get FREE newsletters.
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now!
Hi all,
The multilingual version of DIAX with both IAX and IAX2 support will be
available for download later today or tommorrow.
If someone interested to help me traslate from English to one of the
following languages (ant not oly) please drop me a direct mail.
- italian
- spanish
- french
-
Hey All,
I've started to try and distribute the functionality of my single * server amongst a
few varying servers. The issue I have is that when splitting out the voicemail portion
onto a dedicated server I am no longer able to inform the voicemail application (when
call originated from a
On 28/11/03 07:39, Olle E. Johansson wrote:
The latest version of my Ethereal plugin for IAX2 is now available here:
- http://almaw.com/ethereal-iax2-plugin-0.3.zip
Could you please create a URL that is a bit more non-version-specific?
http://almaw.com/etheral-iax2/
It now, inevitably, has a
Hi,
i've just got 2 grandstream phones and when I try to connect them with *
I get the following:
-- Playing 'demo-abouttotry' (language 'en')
WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call [EMAIL PROTECTED] for
seqno 59134 (Response)
I've seen there
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 02 December 2003 14:03, Alastair Maw wrote:
Could you please create a URL that is a bit more non-version-specific?
http://almaw.com/etheral-iax2/
It now, inevitably, has a web site. :)
Giving a 404. :)
- --
Regards,
Tais M. Hansen
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 02 December 2003 14:03, Alastair Maw wrote:
Could you please create a URL that is a bit more non-version-specific?
http://almaw.com/etheral-iax2/
It now, inevitably, has a web site. :)
Ahh. Typo... Missed an 'e' in the url above.
Lenny,
I've rechecked 0.5.0, but cannot find it in app_dial.c. I found a patch in
Mantis at http://bugs.digium.com/bug_view_page.php?bug_id=366, that
apparently patches app_dial to support the A(x) option. Could it be you're
running an Asterisk that's compiled from CVS, which incorporates the
is there a way for app queue to put into CDR statistics that would
indicate how long the user had waited in the queue?
and is there a way to make app queue to first try to ring the agents and
start music on hold only when they are all talking to other callers?
right now I have it like this
Hi there,
here is my attempt to initiate a restart when convenient from a
software SIP phone.
exten = 588,1,Answer
exten = 588,2,Wait(1)
exten = 588,3,Playback(restart-convenient)
exten = 588,4,Wait(1)
exten = 588,5,Authenticate(0)
exten = 588,6,System(/usr/sbin/asterisk -rx restart when
Try something like this:
exten = 2060,1,Answer
exten = 2060,2,Wait,1
exten = 2060,3,Monitor,wav|algo
exten = 2060,4,Meetme,1|ps
Regards,
Gus
- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 8:58 AM
Subject:
hi,
i am trying to install and see if this works out.
my * is behind a NAT and i have forwarded 4569, 5036
and 5060 ports to the * server which has an private
IP. i am trying to access, authenticate a client who
is outside the NAT and has a public IP. i have
everything configured ok in x-lite
On Mon, 2003-12-01 at 19:19, Bartosz Jozwiak wrote:
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice
lady voice)
And I do not know if it is ok to release it.
What is it written in?
___
Philipp von Klitzing [EMAIL PROTECTED] said:
exten = 588,6,System(/usr/sbin/asterisk -rx restart when convenient)
Put an behind the line?
--
Cees de Groot http://www.tric.nl [EMAIL PROTECTED]
tric, the new way helpdesk/ticketing software, VoIP/CTI,
You can definitely do that with GSM and G.729 when running IAX / IAX2.
Mark
On Tue, 2 Dec 2003 [EMAIL PROTECTED] wrote:
I wonder which voice codec they use, they say one can use a
28k modem using their service which rules out ilbc.
On Mon, 1 Dec 2003 17:34:41 -0500
Chris HARIGA [EMAIL
In C
- Original Message -
From: Roy Sigurd Karlsbakk [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 11:17 AM
Subject: Re: [Asterisk-Users] PREPAID APPLECATION
On Mon, 2003-12-01 at 19:19, Bartosz Jozwiak wrote:
I would like to release prepaid
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, December 02, 2003 7:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dedicated * voicemail server
Hey All,
I've started to try and distribute the
Just to inform the community ... i received an
offer last week for 1 week of asterisk training
+/-2USD !! We can't aford this !
Is that USD 20.000,- as in twenty thousand US dollars, or is have
someone played around with the keyboard? If so - who the fuck can afford
to pay such a
On Tue, 2003-12-02 at 07:44, Philipp von Klitzing wrote:
Hi there,
here is my attempt to initiate a restart when convenient from a
software SIP phone.
exten = 588,1,Answer
exten = 588,2,Wait(1)
exten = 588,3,Playback(restart-convenient)
exten = 588,4,Wait(1)
exten =
It is a shame that within a couple of hours they can tell you to remove helpfull
documentation, but not (seemingly) help answer questions regarding there Cisco stuff
on this list. I think Cisco must have their priorities mixed up!
Just my opinion... which also means I won't support a company
You could add an initial digit based on whether it was a busy or no
answer forward, use the extra digit to determine the message played on
the VM server and just strip it back off to get the mailbox number.
Email me direct if that isn't clear enough.
This is actually what I have at the
Hi Gus,
Thanks. It worked
regards...
Girish
From: CW_ASN - Gus [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Meetme Recording
Date: Tue, 2 Dec 2003 10:56:18 -0300
Try something like this:
exten = 2060,1,Answer
exten = 2060,2,Wait,1
I'm a windows developer ready to develop TSP/MSP (TAPI) drivers for
asterisk PBX but i searched a little more support to develop these
drivers ...
unfortunatly i have to develop the drivers commercially because i will need
to hire a asterisk freak to explain me in detail how everything works
I get echo on my X100P for about 15 seconds then it disappears. Believe me,
when I get a chance I'm going to tweak it since my wife nags me about it. I
converted my house over to *.
My setup:
Compaq Deskpro EN SFF P3 500
Dual NIC for real IP and internal IP
X100P Card for PSTN.
Cisco ATA186
This is also in production use
http://www.tyan.com/products/html/gx15b2723t15.html
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
It is a shame that within a couple of hours they can tell you to remove
helpfull documentation, but not (seemingly) help answer questions
regarding there Cisco stuff on this list. I think Cisco must have their
priorities mixed up!
Just my opinion... which also means I won't support a company
Must have included a week in Amsterdam
Scott M. Stingel
Emerging Voice Technology Inc.
URL:www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roy Sigurd Karlsbakk
Sent: Tuesday, December 02, 2003 2:23 PM
To:
You hear it very well !
As i think i'm polite, so i'm not going to put the name of the company online but
believe me their was a hole in the seiling after i read the email ...
Van: Roy Sigurd Karlsbakk [mailto:[EMAIL PROTECTED]
Verzonden: di
Roy Sigurd Karlsbakk wrote:
Just to inform the community ... i received an
offer last week for 1 week of asterisk training
+/-2USD !! We can't aford this !
Is that USD 20.000,- as in twenty thousand US dollars, or is have
someone played around with the keyboard? If so - who the
At 9:27 AM -0500 12/2/03, Richard Alexander wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, December 02, 2003 7:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dedicated * voicemail server
Hey
Just to inform the community ... i received an
offer last week for 1 week of asterisk training
+/-2USD !! We can't aford this !
Is that USD 20.000,- as in twenty thousand US dollars, or is have
someone played around with the keyboard? If so - who the fuck can afford
to pay
the details :
16000USD for the training
+ 10% for administration
+ Travel costs to Belgium (from canada)
+ Hotel costs ...
so 'im not far from 2USD ...( it could be even more )
We have a budget but not such a big one ... and even if i would, i would be a bad
manager to accept such a
maybe it means United States Dimes :) $2,000 ain't bad for a week of
training.
But to answer your question, I have a friend that does Checkpoint firewall
training/consultation and he gets upto $20,000 per week for running training
classes. Not in the US mind you but abroad, mostly in Europe. He
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, December 02, 2003 9:47 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Dedicated * voicemail server
You could add an initial digit based on
Hello,
Right now I have app queue working with incominglimit=1, there is no
call waiting signal, but after a while( like couple of hours) some
phones randomly get stuck. The * thinks that they are in use and doesnt
ring them, when they are infact not in use.
sip show inuse, shows that they are
I see that when an Asterisk connects to another one via IAX, it seems to
use port 4569 for the first one. But if it has multiple IAX connections
the additional ports seem to be chosen at random.
Is there anyway to predict, or specify which ports or range of ports to
use, for the sake of
American companies are too cheap? That's laughable. I could get into
economic's but this is not the place. Your typical Cisco firewall PIX
training class avgs. $2499 a week. I can only think he's training 10 people
@ $2,000 a week. If he's making $20k for a week of training, no wonder why
Hello,
Right now I have app queue working with incominglimit=1, there is no
call waiting signal, but after a while( like couple of hours) some
phones randomly get stuck. The * thinks that they are in use and doesnt
ring them, when they are infact not in use.
sip show inuse, shows that they are
You mean on US or EU dialup, but I doubt you will get any success on far
east dialup or african dialup with that. Here you would either need
speex or g723.1.
Mark Spencer wrote:
You can definitely do that with GSM and G.729 when running IAX / IAX2.
Mark
On Tue, 2 Dec 2003 [EMAIL PROTECTED]
We are a little Belgian company working for the automotive world (car builders,
leasing companies , ...) and believe me if i make an offer of 2USD for a week i'll
never have to make an offer again !
I also worked in the past (2000 which is considered as the craziest year for Tech
Hi!
exten = 588,1,Answer
exten = 588,2,Wait(1)
exten = 588,3,Playback(restart-convenient)
exten = 588,4,Wait(1)
exten = 588,5,Authenticate(0)
exten = 588,6,System(/usr/sbin/asterisk -rx restart when convenient)
exten = 588,7,Hangup
The problem: We never reach the convenient
what do the options algo do in the monitor app? I dont see that in the show
application monitor? is this a patch?
Dave
[EMAIL PROTECTED] 12/2/2003 6:56:18 AM
Try something like this:
exten = 2060,1,Answer
exten = 2060,2,Wait,1
exten = 2060,3,Monitor,wav|algo
exten = 2060,4,Meetme,1|ps
Hi list.
I'm having the next problem.
I have a * with 1 TDM400P (4 ports) and
one X100P, with a working configuration.
Today i add one more X100P card, and i change
the config files as next:
zapatel.conf:
fxsks=1-2
fxoks=3-6
loadzone = us
I've seen this same thing. But it doesn't happen only for phones using
the queue I believe it is a bug in the chan_sip driver. What I have
found is that when a phone sip phone is unplugged/not registered and a
call comes in it increments the counter and doesn't reset the counter
when the phone
Hi,
The PBX at the Colorado Organization for Victims' Assistance fried as a
result of the building power being cycled. I'm now in the process of
building an * system to replace the failed PBX. Minimum cost is the
priority.
I have a T100P card installed in the new system, and I am about to
After battling for days trying to figure
out whatwas wrong with my iax.conf it was determined that I do not have
any inkeys set on the digium server. Now whether that is something new or
just in a few cases I am not sure. Messing around and reading on IRC and
the mailing list I could get
- Original Message -
From: Mark Spencer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 3:22 PM
Subject: Re: [Asterisk-Users] VoiceGlo
You can definitely do that with GSM and G.729 when running IAX / IAX2.
Mark
On Tue, 2 Dec 2003 [EMAIL PROTECTED] wrote:
No takers? Should I submit a bug report then? I didn't find any open
bugs on stuck
MWI.
Clif Jones wrote:
I have had several cases where the message waiting indicator was stuck
in the on state
with Cisco 7960 SIP phones. Here are the two cases:
1. Single extension that mapped to a single
I think we need to look carefully at the situation here. We all know that
training on complex specialist products can often come in at between
2-4,000USD per week per individual (certainly this is my commercial
experience). However normally when I go on a training course there are
around 10
On 02/12/03 16:32, Matt Lawson wrote:
I see that when an Asterisk connects to another one via IAX, it seems to
use port 4569 for the first one. But if it has multiple IAX connections
the additional ports seem to be chosen at random.
Is there anyway to predict, or specify which ports or range
Hi,
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to the
Call Manager I released last week) and it seems to regularly miss
incoming calls entirely. It also occasionally misses the drop signal
when the remote
I am having problems in a couple of installations where I have SIP
phones (both GS101 and ATA186) connecting to an asterisk box that has a
public IP address, where the stations are behind NAT.
I'm still testing to make sure I have all the permutations looked at,
but from what I can tell, what
WROOGGG
Voiceglo's webphone is IAX and they use GSM. I have my Asterisk server
registered with voiceglo right now.. so I know for a fact its IAX :P
s you didn't hear that from me.
bkw
On Tue, 2 Dec 2003, Adam Hart wrote:
Does asterisk support G.729a or do you have to add
something (is there an open source one)
Todd Wallace
Mark,
Following email of a couple of days ago - if you could confirm if you want
me to put things together for you in Paris. As mentioned I am in London from
24th to 30th but otherwise Paris.
Stephen
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday,
Title: Mensaje
Hi, I want to use
G.723.1 on *, I read it is supported in Pass Through mode, but I don't
understand whats the meaning of that.
I have a GW 5300 and
an ATA 186 and I want to place calls to PSTN.
I setup this
config:
[general]port =
5060
bindaddr =
xx.xx.xx.xx
context =
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 12:36 PM
Subject: [Asterisk-Users] Configuring new system for a non-profit organization
Hi,
The PBX at the Colorado Organization for Victims' Assistance fried as a
- Original Message -
From: Matt Lawson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 11:32 AM
Subject: [Asterisk-Users] IAX port numbers?
I see that when an Asterisk connects to another one via IAX, it seems to
use port 4569 for the first one. But if it has
Sorry, I dont know but i want a term added for archive searchablilty.
flex grow flexgrow flex-grow
- Original Message -
From: Michael Welter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 12:36 PM
Subject: [Asterisk-Users] Configuring new system for a
I am curious if anyone has tested, or is using any of the ArtDio gateways?
http://www.artdioinc.com/eng/product.htm (flash site)
Do they function as I might be construing them?
The IPS-1101 is listed as having 1 FXS and 1 FXO. Does that mean it will
actually control two seperate calls, one on
bottom response = on
On Tue, 2003-12-02 at 11:36, Michael Welter wrote:
Hi,
The PBX at the Colorado Organization for Victims' Assistance fried as a
result of the building power being cycled. I'm now in the process of
building an * system to replace the failed PBX. Minimum cost is the
I have asterisk boxes in 2 different buildings each connected to the telco
with a PRI. I am now setting up asterisk machines in remote buildings -
dialing out via one of the other 2 machines. These are a snip from each
extension.conf on 1 remote and the 2 machines connected to the PRIs, to
On Tue, 2 Dec 2003 12:14:24 -0600, Steven Sokol [EMAIL PROTECTED]
wrote:
Hi,
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to the
Call Manager I released last week) and it seems to regularly miss
incoming
Michael Bielicki wrote:
You mean on US or EU dialup, but I doubt you will get any success on
far
east dialup or african dialup with that. Here you would either need
speex or g723.1.
Mark Spencer wrote:
You can definitely do that with GSM and G.729 when running IAX /
IAX2.
Mark
yes, * supports it.
but is available only by commercial license.
10$ / channel directly from digium.
Matteo.
Il mar, 2003-12-02 alle 19:37, Todd Wallace ha scritto:
Does asterisk support G.729a or do you have to add something (is there
an open source one)
Todd Wallace
--
Brancaleoni
Alastair Maw wrote:
On 28/11/03 07:39, Olle E. Johansson wrote:
The latest version of my Ethereal plugin for IAX2 is now available here:
- http://almaw.com/ethereal-iax2-plugin-0.3.zip
Could you please create a URL that is a bit more non-version-specific?
http://almaw.com/etheral-iax2/
It now,
At 10:37 AM 12/2/2003, you wrote:
Does
asterisk support G.729a or do you have to add something (is there an open
source one)
Yes, Yes, and Maybe (i.e. it's not free, but you can license one through
Digium, and there is a reference source available but absolutely NOT
open-source).
Check out this
Steven Critchfield wrote:
On Tue, 2003-12-02 at 07:44, Philipp von Klitzing wrote:
BTW: Where exactly is the difference between Hangup and Softhangup()?
Hangup is something done in the course of the dialplan and works on the
current channel where softhangup is a cli command that works on a
You can buy g729 lic from digium for 10.00 per channel.
bkw
On Tue, 2 Dec 2003, Todd Wallace wrote:
Does asterisk support G.729a or do you have to add something (is there an open
source one)
Todd Wallace
___
Asterisk-Users mailing list
[EMAIL
I am having problems in a couple of installations where I have SIP
phones (both GS101 and ATA186) connecting to an asterisk box that has a
public IP address, where the stations are behind NAT.
I'm still testing to make sure I have all the permutations looked at,
but from what I can tell,
Title: Message
just
need to buy g729 licences from www.digium.com, install it and off you go.
:)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
WallaceSent: Tuesday, December 02, 2003 6:37 PMTo:
[EMAIL PROTECTED]Subject:
Philipp von Klitzing [EMAIL PROTECTED] said:
exten = 588,6,System(/usr/sbin/asterisk -rx restart when convenient)
Put an behind the line?
It does help to get a proper hang up for the client, but there is no
restart initiated at all... looks like now the system calls gets
cancelled due to
Title: 'Stop Now', 'Restart' problems
I'm not sure where to start looking for a solution on this. I use use Asterisk::Manager to reload Asterisk with a command like:
$astman-sendcommand( Action ="" 'Command', Command = 'Reload' );
After a while, when I try to do a manual restart or 'stop
Steven Sokol wrote:
Hi,
I seem to be having problems with IAX clients based on the iaxClient
library. I have been working on my own client (an augmentation to
the Call Manager I released last week) and it seems to regularly miss
incoming calls entirely. It also occasionally misses the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
Sent: vrijdag 28 november 2003 5:11
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk behind NAT How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented
I would stay away. I have evaluated these units and returned them. I
determined that this
unit or one from them that fits this description was actually a unit
that you put between
your phone and your phone line (1 FXS 1 FXO claim) and hook to the
ethernet. This
unit would connect the FXS
I don't think so, is it zapata.conf , echotraining
= yes
Please confirm and I will try it
Rob
- Original Message -
From:
TC
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 7:01
PM
Subject: Re: [Asterisk-Users] Tone
Detection Problem
is echo
Dan wrote:
Hi all,
The multilingual version of DIAX with both IAX and IAX2 support will be
available for download later today or tommorrow.
If someone interested to help me traslate from English to one of the
following languages (ant not oly) please drop me a direct mail.
- italian
- spanish
-
I've setup asterisk on a Dell Poweredge 1300 Server. PIII 600, 500mb
Ram, SCSI HD's. It is running Fedora Core 1. I have 3 X100P cards and 6
Grandstream IP phones.
Everything works but the sound quality with the FXO cards is poor. Static and
choppiness. It works great between the IP phones.
The
Sorry to everyone on the list, but for some reason
this is the only reliable way to get hold of John.
John Brown of Chagres Technologies, please contact
me! I have been trying for weeks now to get hold of you via email and
phone after wire transfering money into your account for the
On Tue, 2 Dec 2003, tony banks wrote:
I have 1 IP 7940 with the following Firmware versions
App Load ID:
P00303011201
Boot Load ID:
PCO303010001
Version
3.1(12.1)
Sounds like a Skinny image to me (and an old one, around CCM 3.1, too.)
You can either get a SIP image from cisco and
John,
I have 12 7960 phones with 6.0 with no issues. Sounds like a hardware problem to me.
-Original Message-
From: John Todd [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 30, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 6.0 + Asterisk question
I have
Michael,
Where in your extension definition to you dial a channel (SIP, Zap, or other)? You
are missing the dial entry.
-sb
-Original Message-
From: Lists [mailto:[EMAIL PROTECTED]
Sent: Saturday, November 29, 2003 10:53 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Sip Issue
Hi Todd,
Yes Asterisk supports G.729a, you can by licenses
for it at www.digium.com .
I think there is a free to use one, but only for
windows, * your best off with the digium licenses.
Greetings,
Tjardick
- Original Message -
From:
Todd Wallace
To: [EMAIL PROTECTED]
Please
Remove me from the list
ReggieReginald Huey
Do you Yahoo!?
Free Pop-Up Blocker - Get it now
What they are probably marketing is putting in their own equipment out
there. I install a product that does exactly that. A paradyne jet
fusion. It takes care of the part of which channels are data and which
are voice.
If it's anything like these, the lines will come out on pairs. You will
Hi robert,
I found that the disallow=all and then specify a codec with allow= was required in
sip.conf.
[17471234567]
type=friend
username=17471234567
secret=censored
host=dynamic
nat=yes
disallow=all
allow=ulaw
Jon Hopper
robert ivanc [EMAIL PROTECTED] wrote ..
Hi,
i've just got 2
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 2:14 PM
Subject: RE: [Asterisk-Users] VoiceGlo
Hi,
Anyone knows what USB phone are they using? Where can one get it from?
algo is a file where app write a wav data. In spanish, algo means
something... :)
Gus
-= Info about application 'Monitor' =-
[Synopsis]:
Monitor a channel
[Description]:
Monitor
Used to start monitoring a channel. The channel's input and output
voice packets are logged to files until the
On Tue, 2003-12-02 at 12:55, Howard White wrote:
bottom response = on
The magic question to ask CBeyond is whether the T1 they provide you is
Primary Rate Interface (PRI) or Basic Rate Interface (BRI). Their web
site is too heavy on pretty marketing and wy short on technical
details.
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