James, it's a piece of cake, you should be able to do this in an
afternoon with about the same for the billing app.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Sunday, March 13, 2005 12:15 PM
To: Asterisk Users Mailing List -
Taking yourself off mute is one of the more important requirements for
broadcast conferences.
I probably dial in to about 3 conference calls a week (using commercial
services) where the default is everyone in the call is on mute and then
you press star to talk - some automatically take you off or
Title: AntiSpam Alert: Request For Authentication
Is there anyway we can get this shit off
the asterisk list apart from posting their email address [EMAIL PROTECTED] here for the spambots
to pick up?
Dean
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday,
Adam,
I can only comment on Australia but you list one.tel as one of the
companies you can deliver sms's to.
One.tel ceased operations at least 18 months ago if not more.
I'm glad you developed this product and think any external asp delivered
service for asterisk is an exciting development but
And it will give you more flexibility for future development.
Having sold pabx/call centre technology in the past I'm still blown away
with how good Asterisk is on an even price basis, the fact that Asterisk
is 1/3rd at the most the price leaves everything else for dead.
Cheers,
Dean
Nope that wont work.
The whole reason you need to change and unchanged mute for conference
calls is when on a cell phone or listening in on 'broadcast' conference
where the majority of the time I will be just listening in eg - a
brokers report or similar but I may want to ask a question at the
Where did you get 1.05.23 from? The doc is available on the grandstream
site but not the actual firmware.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Tuesday, March 15, 2005 11:48 PM
To: asterisk-users@lists.digium.com
Subject: Re:
I'm using faktortel.com.au so can email you configs if you thing it
might help.
(BTW I think they all use the same gateway and are just resellers).
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Davidson
Sent: Wednesday, March 16,
There is a script on the [EMAIL PROTECTED] sourceforge list that reads the
weather for you.
Basically ftp's a text file from the BOM and then uses festival to read
it out to you
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Just use [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/
Meetme2 is automatically installed
Cheers
dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anil Kumar
K
Sent: Friday, March 18, 2005 10:56 AM
To: Giovanni Powell
Cc:
Discussion
Subject: Re: [Asterisk-Users] About the weather..
dean collins wrote:
There is a script on the [EMAIL PROTECTED] sourceforge list that reads the
weather for you.
Basically ftp's a text file from the BOM and then uses festival to
read
it out to you
:)
Yeah but he doesn't want to use
Thanks for the replies though.
Kris
dean collins wrote:
Lol, if he could do all that then he wouldn't need the [EMAIL PROTECTED]
weather script.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Friday, March
Do a search on Empirix.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, March 19, 2005 1:39 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk
I?m a telecommunication engineering
There was a big discussions at the Future of IP Heavyreading.com
conference in NY this week about this.
It's not as easy for the ISP's to get away with it as you think.
Empirix.com were called in to investigate and document the recent Vonage
case (lol interestingly enough they have also been
What about a Sony U71, again not exactly pock friendly but would make a
hell of a SIP terminal.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris
Edwards
Sent: Saturday, March 19, 2005 4:33 PM
To: Asterisk Users Mailing List - Non-Commercial
Great, just received 9 virus emails in the past 24 hours
from the asterisk list where people have had my address in their address book.
Heads up people, its an attachment, the text looks a
little jinglish why would you open it?
Cheers,
Dean
Peter,
You need to spend some more time reading the wiki, your question is far
to basic for someone who had invested anything more than about 15
minutes looking at asterisk.
http://www.voip-info.org/wiki-VOIP+Service+Providers+Residential
this is a basic list of voip providers some will
Take a look on the [EMAIL PROTECTED] sourceforge forum for festival-weather,
it's like the most popular topic over there.
I think like about 50 people have downloaded it since it got put up a
few weeks ago.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
: dean collins [mailto:[EMAIL PROTECTED]
Sent: Sunday, March 20, 2005 7:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] virus
Great, just received 9 virus emails in the past 24 hours from the
asterisk list where people have had my address in their address
: Friday, March 18, 2005 11:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] virus
On Sun, 2005-03-20 at 08:17 -0500, dean collins wrote:
Great, just received 9 virus emails in the past 24 hours from the
asterisk list where people have had my address
otherwise be great.
Thanks to everybody for your replies...great info!
-Pete
On Mar 20, 2005, at 5:22 AM, dean collins wrote:
Peter,
You need to spend some more time reading the wiki, your question is
far
to basic for someone who had invested anything more than about 15
minutes looking
Yep :)
Use a grandstream and [EMAIL PROTECTED] and you only need to push a single
button and go straight through to messages.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Charlton
Sent: Monday, March 21, 2005 11:18 AM
To:
Already available using group-settings in AMP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Shaw
Sent: Monday, March 21, 2005 9:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] [EMAIL PROTECTED] .6
Brian, are you sure you cant port your number aware from Vonage?
I was at a VOIP conference last week and the Vonage spokesman said quite
clearly that Vonage allowed people to port numbers out - although he did
say that not all carriers were set up to receive them.
Cheers,
Dean
-Original
Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/
Or directly from Grandstream at
http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip
Release notes doc here
http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc
while on the matter I just want to extend a
Type help-ahh from the console and you
will be able to change logins etc.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, March 25, 2005 10:08
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] What
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20door
gee that took a lot of effort.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, March 25, 2005 11:57
AM
To:
asterisk-users@lists.digium.com
Subject:
Jermal,
Your second round of questions are just as basic, do some research on
wiki.
1/ did you not see that when you log onto the console it says type
help-aah to change passwords?
2/ [EMAIL PROTECTED] doesn't need any more documentation - all of the
documentation for [EMAIL PROTECTED] is on the
Discussion
Subject: CAUTION: Re: [Asterisk-Users] grandstream firmware update
1.0.5.23
CAUTION: voicemail screwed up for me (garbled) with upgrade to 23, went
back to .22 and all is well.
Don't know why, I'll look at it later.
dean collins wrote:
Version 1.0.5.23 is now available from http://gs
Noah you can, why not use amp (via [EMAIL PROTECTED]) to configure incoming
groups.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Silverman
Sent: Friday, March 25, 2005 1:01 PM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Two
download it separately?
-N
dean collins wrote:
Noah you can, why not use amp (via [EMAIL PROTECTED]) to configure
incoming
groups.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Silverman
Sent: Friday, March 25, 2005 1:01 PM
To: Asterisk
Because you learn how asterisk works by looking at the AMP dial plan
fast.
With phpconfig you have the ability to view each of the aspects and
learn a lot faster than reading the wiki.
It also gives you an understanding on how a professional dial plan
should be laid out.
If he doesn't even
I've made a lot of changes to the AMP dial plan including implementing
the asterisk weather script.
[EMAIL PROTECTED] is a great building block to start from but it's not
restricting anything you want to implement later.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
I worked on a project for Vodafone Australia that did exactly this.
You sms'd a list of phone numbers to an automated 'conference number'
60 seconds later it called your phone back, and then all of the other
numbers adding them into the conference call successively.
I thought it was a great
Well not entirely skype has had conferencing for a long time and they
also have run beta trials on skype voicemail but have chosen not to
implement it commercially yet.
I don't see the point of this discussion.
Skype is the pstn service;
Asterisk is a pabx;
No one ran around telling ma bell she
Hi Nitesh,
Take a look at this
http://www.microappliances.com/site/html/index.php?section=Productspage
=clienthowto.php
I've never implemented it though so I would appreciate some feedback on
if it works.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi Greg,
I think he was talking about creating a call via the website not
actually dialing a phone number.
Having said that can you post your script here, I'd like to see it.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor
Mark, what exactly are the limitations you are finding?
You do know that you can make modifications to the AMP dial plan don't
you?
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Charlton
Sent: Tuesday, March 29, 2005 6:16 PM
To: 'Asterisk
Sure Robert, but you are going to get a lot of cross posted questions
for certain topics (though I agree this one was totally about
[EMAIL PROTECTED] so should have gone to the sourceforge forum).
Post here if it is a straight asterisk question but on sourceforge
my PSTN
linked in, and then probably a cheap SIP / IAX hardphone.
Cheers
Mark Charlton
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: 30 March 2005 00:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
Ron, can I suggest a little more research next time. Everything you are
asking is already very well documented on the wiki.
The answer to your question is Yes - Asterisk can do call conferencing.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
Cheers,
Dean
-Original Message-
From:
Yep I agree with you chuck. I'd much prefer a web based forum.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Thursday, March 31, 2005 10:27 AM
To: Linux - PBX, Asterisk
Subject: [Asterisk-Users] Are there online forums instead of this
Err - not in Australia at the moment but all of Australia uses ETSI -
BRI
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, March 31, 2005 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thomas, can I suggest you do a little more
research before posting your question to this list. The wiki has all of the
information you are looking for.
http://www.voip-info.org/wiki-VoIP+Phones
Any SIP or IAX handset/ATA will work with
Asterisk.
Cheers,
Dean
Hi John,
Not sure exactly what you are planning to offer but be aware I've been
using www.faktortel.com.au for about 3 months now.
Sensational and almost flawless service.
Unlimited incoming calls to my Sydney 02 number that gets diverted to my
asterisk server here in New York for only A$12 a
Thats correct, you cant use the
backup to jump revisions.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Tuesday, April 05, 2005 1:37
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] AAH
Not much to configure you just need a way for people to access the
voicemail extension.
I use [EMAIL PROTECTED] and in the initial IVR I have a hidden option to
dial the *98 extension.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Lol - Cisco for one.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matteo
Brancaleoni
Sent: Thursday, April 07, 2005 7:12 AM
To: [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Sangoma VS.
Magnus Are you freaking psychic?
You are not going to believe this but I just finished a phone call 5
minutes ago with a California based developer who has a proprietary
version of this already available and is looking to share this with the
asterisk community.
His does even
Thanks Robert, that information is really handy to know - now turn the
freaking out of office message off!
-Original Message-
From: Huddleston, Robert [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 07, 2005 4:57 PM
To: dean collins
Subject: Out of Office AutoReply: [Asterisk-Users
That's not what he is looking for though, this will only dial 1 number
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josiah
Bryan
Sent: Thursday, April 07, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
As long as this competition is setup to be delivered after July I should
be able to donate a www.akimbo.com box to the winner (I'm going to be
distributing them in Australia).
You will still need to pay for the content and the monthly charges but
it's still a kick arse prize.
Cheers,
Dean
Packet 8 $19.95 (but I use the $49 international plan)
Connects straight in via ata and x100p.
USA quality is vgood, international could be better from time to time
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
I don't understand why you wouldn't just do this via contexts and manage
the trunk dial plan on your existing pabx?
Am I missing something?
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jason Kawakami
Sent: Friday, April 08,
I have a similar situation but it seems to vary from call to call
sometimes.
Using 2 digium genuine x100p's in a dell with riser card.
I'm wondering if it is something to do with the riser because it doesn't
seem to matter if I swap various cords, positions, etc.
Cheers,
Dean
EUREKA
I finally solved this problem, I dont know why some of the
more experienced people in here haven't answered this question (I guess they
dont use Stanaphone but here it is)
The problem isn't in how you register with Stanaphone but with
the AMP config :(
in the sip.conf
What country are you in, and does the chipset on the compat card
support the telco standards in your country?
If the chipset doesn't match your telco standards, there is a high
probability you won't get rid of the echo. If it does match, then try
echotraining=800
echocancel=yes
[DC]
Well mine is legitimate digium
And I'm in the usa
Here is the output but I have no idea what that means?
[EMAIL PROTECTED] root]# cat /proc/interrupts
CPU0
0: 490763 XT-PIC timer
1: 2 XT-PIC keyboard
2: 0
Lol, just posted a question to the list that should keep away any
bidders.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Charles Osstyn
Sent: Monday, April 11, 2005 4:01 PM
To: asterisk-users@lists.digium.com
Subject:
Title: SIP ACD system for station to station calls
Colin,
Asterisk does exactly what you are after,
as for your comment about high availability it can be.
Either research or hire in the expertise
your requirements are already being met in a number of very similar
installations.
I'd also like to hear the answer, I probably will never have a need for
the information but you never know.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian Capouch
Sent: Wednesday, April 13, 2005 6:13 PM
To:
Ben probably better to post to the [EMAIL PROTECTED] sourcforge list seeing
this is a specific [EMAIL PROTECTED] problem, having said that please post a
console
output of the problem occurring.
If you don't know how to do this please email me and I'll explain how to
use putty to view the
Lol, Nevada isn't all fly by night, that's almost as saying Delaware is
a haven for injunction protection seeking companies like Microsoft.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Thursday, December 30, 2004 4:25 PM
Jeff,
You need to install using the linux 'expert' command at the beginning of
the installation
Basically I had the same problem.
Some nic cards are not recognized by xorcom and therefore don't request
dhcp etc.
Michael,
I was never able to upload my old configs onto Xorcom, I hope you have
Hi Benoit,
Have you actually implemented an asterisk installation?
Can you provide more information about how this works?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of B. Vallet -
www.acropolistelecom.net
Sent: Wednesday, January 05,
Yep check out the new generation of set top boxes - all ip based.
eg www.akimbo.com just launched at CES yesterday, both Ethernet cat 5
and wireless connections.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Graves
Sent: Friday,
Hey I noticed this posting, is anyone in New York interested in catching
up?
I'd be happy to host it at my place on 72nd/york if it wasn't too big a
group, or we can always head out and grab some lunch or something
somewhere.
Email me your interest and we'll see what the numbers are.
Cheers,
Can anyone comment on Broadvoice call quality? Particularly calls
from the USA to Asia and Australia.
I currently use Packet 8 and am contemplating changing over.
Cheers,
Dean
___
Asterisk-Users mailing list
Ha ha ha you are kidding right Chris? Americans are the most arrogant
people on the face of the planet.
Lets not make this a 'holy war' about nationalities we all have our own
shortcomings (like aussies who would prefer to sink piss (beer) rather
than expend our energies in more significant
Ronald, it's the context listed in voicemail.conf (I got caught on this
as well)
I really wish Asterisk was better documented; it's bullshit the way it
stands at the moment.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Capouch
Sent: Thursday, January 13, 2005 2:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Grandstream Bugetone 101 mwi
dean collins wrote:
Ronald
-Users] Grandstream Bugetone 101 mwi
dean collins wrote:
In addition I currently have a $2,000 bounty for a video version of
Meetme and haven't had a single person even reply to take up the
challenge.
How much more encouragement do you need?
One of the hardest things for traditional
Do a search on ACD and agents, this is certainly achievable.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis
Voitenko
Sent: Friday, January 14, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Routing
Jonathan,
Log on to [EMAIL PROTECTED] with user name :root password: password
Then run: netconfig
You'll probably find your nic wasn't automatically assigned an IP
address.
You will need to enter in all of the details manually.
Cheers,
Dean
-Original Message-
From: [EMAIL
Breaking News - Powell
resigns as FCC Chair
Federal Communications Commission (FCC)
Chairman Michael Powell announced his resignation today (January 21, 2005) and
will step down in March. In a brief statement on the FCC's website (www.fcc.gov) Powell
said, Having completed a bold and
There are heaps but why not use a headset
If you insist on usb handset then there are 3 listed here.
http://www.telecoms.co.uk/catalog/default.php?cPath=583_829_830
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adi Linden
Sent: Sunday, January 23,
Im using tftp server that automatically loads on each
reboot, for some reason the last 2 files fail to load each time. (and I think
this has always been the case)
Aborted 192.168.16.32 C:\Program Files\TFTP
Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25
Jan 18:25 Error
Hi Jason,
Doesn't sounds like it.
I run a P3 -750 and takes about 15 minutes to install.
I don't have advice on what else you can do. Just letting you know a P@
400 might be a little underpowered but should work.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Im trying to install [EMAIL PROTECTED], Ive just
downloaded the latest cd from soundforge. I can get it to install ok (network
card didnt auto configure but I worked out how to use netconfig).
I worked out how to add a few grandstream budgetone fine. Worked
out how to upload music etc.
I'm not sure if this will work with your cisco's but I can guarantee
that it works with the grandstreams.
This is what I use to update my 4 phones, running on my main winxp
machine and it's free for non commercial use.
http://www.weird-solutions.com/product/tftpc2000.html
Cheers,
Dean
Sorry my mistake, wrong link, here is the correct one.
http://www.weird-solutions.com/product/tftp-desktop.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dean
collins
Sent: Wednesday, January 26, 2005 9:13 AM
To: Asterisk Users Mailing List - Non
I posted this question on the AMP forum but there doesnt
seem to be much traffic on their forums, is there anyone on this list that can
help me out. Going crazy not being able to get this working.
Cheers,
Dean
By: dean collins - deancollins
IAX2 trunk question
.
Remember I'm very new at this, but I didn't see anyone
respond to your post.
Goog luck, David
- Original Message -
From: dean collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, January
26, 2005 5:36
).
--
Dana
- Original Message -
From: dean collins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 25, 2005 6:28 PM
Subject: [Asterisk-Users] grandstream budgetone-100 updates
I'm using
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4
Can I make a suggestion that some documentation is provided
for the Tortoise CVS download of the asterisk docs. Ive tried every
combination and I cant get it to work.
Im assuming it must work otherwise it wouldnt
have
?
Cheers,
Dean
-Original Message-
From: dean collins
Sent: Friday, January 28, 2005 10:56 AM
To: 'andrew'
Subject: RE: FAQ missing info
Btw, do you need the pstn line for the X100P plugged in while running
the install?
Just downloaded and burning the cd now.
-Original Message-
From
HI, not essential but is there a bug list for [EMAIL PROTECTED]
If so the automatically generated URL for AMP voicemails isnt working
ext 30,
There is a new voicemail
in mailbox 30:
From: ext
32 32
Length: 0:07
seconds
Date: Friday,
January 28, 2005 at 03:41:50 PM
Dial
) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/30-5dde' i n macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/30-5dde' asterisk1*CLI
any thoughts?
Cheers,
Dean
-Original Message-
From: dean collins
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of dean
collins
Sent: Friday, January 28, 2005 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FW: FAQ missing info? [EMAIL PROTECTED] V 0.4
Yes, now runs an updated version, also has MeetMe2 which
Can anyone tell me when Meetme2 is scheduled for release?
http://www.voip-info.org/wiki-Asterisk+MeetMe2+Design
Cheers,
Dean
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Asterisk-Users mailing list
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the command
line cvs application that comes with cygwin (www.cygwin.com) and follow
the command line instructions provided. This works even better when you
want to sync docs and code all the time.
Cheers,
Guills
From: dean collins [mailto:[EMAIL PROTECTED
download for Asterisk Docs
On Sat, 29 Jan 2005 11:10:25 -0500, dean collins [EMAIL PROTECTED]
wrote:
What should I be using to read this so it doesn't show the XML?
This would have been more appropriate to post to the asterisk-docs
mailing list, but either way.
The documentation is written
Hi, not sure if it is against the rules
to sell second hand equipment in here but havent seen anything against
it so here it is.
Im upgrading to 2 lines so I have
some spare equipment for sale here. This is an ideal starter pack and will get
you going with 1 line and 1 extension.
1
Yep, basically it is a SIP video phone like a grandstream is a SIP voice
phone.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ferguson,
Michael
Sent: Saturday, January 29, 2005 6:44 PM
To: Ing. Ignacio Ortega A.; Asterisk Users Mailing
Why does the X100P have echo and the Wildcard TDM400P have
no echo?
I thought the only advantage of using the TDM400P was that
it used less interrupts than the X100P?
Are there any other advantages?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
: [Asterisk-Users] Re: grandstream budgetone-100 updates
dean collins wrote:
I'm using tftp server that automatically loads on each reboot, for
some
reason the last 2 files fail to load each time. (and I think this has
always been the case)
Aborted 192.168.16.32C
http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20Me%20video%20conferencing
I posted this bounty for $US2,000 some months ago.
Basically I needed the ability for 4 or 5 of us to
conference on a weekly basis which is why I was happy to offer this bounty,
however
, February 02, 2005 9:31 AM
To: dean collins
Subject: Re: [Asterisk-Users] 403 forbidden error
Thanks alot man...
No, actually I dont know what [EMAIL PROTECTED] is? im going to search
online for it now but if you wouldnt mind sending me the link that would
be great.
Thanks again
ken
dean
You don't need to enable it. It happens automatically.
Log into console
With username : root
Password :password
then enter asterisk -r
To get out type exit.
What happens when you enter asterisk -r?
If you type sip show peers what happens?
-Original Message-
From: [EMAIL PROTECTED]
According to Volume 1 Asterisk Docs
To create an FXO channel on the same TDM400P card, we list all the
settings for the channel and then define the channel number. Instead of
only having signalling be fxo_ks though we want the signalling to be
fxs_ks. Because the other settings haven't been
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