If your working with Virtual PBX then why not set your users with there own
rules and normal extension numbers in there own context. You can have many
context.
That way only extensions you allow to see the context there in will have
those options.
- Original Message -
From: Marcello
Justin Johnson wrote:
Hi All,
I have centOS 4.3 installed and have attempted to install asterisk
separately. I have installed all the modules as suggested on Asterisk
downloads, more (via SVN) However, on the zaptel install I am getting
the following errors.
centosbug is, like, a problem
John Kington wrote:
I tried to get an update from NuFone but
Has anyone gotten their tollfree number ported
to another provider by NuFone? Should I just
forget it and move on?
Regards,
John
Yes we have ported our number out of there service. You need to go and sign
some papers with the
Rich Adamson wrote:
Has anyone attempted to use FreePBX for a business in production mode?
Yes it works great in business applications.
Initial take is there are lots of things scripted but a lot of
limitations in terms of supporting basic business functions. Inability
(or lack of
I normally don't like talking bad about products. But I would like to say
that the Welltech/Wellgate are not products that are support to work with
asterisk. I have invested many hours of work in getting there device to
work with Asterisk. They do not. And also as of Last Nov. They told me
Iaxtel has been down for some time now.
But to get in contact with digium via your asterisk box all you need is to
set this dialing rule up.
exten = 500,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ;Call Digium
exten = 500,2,Congestion
Kerry Garrison wrote:
Is IAXTEL still around? I
Ken D'Ambrosio wrote:
There are a bazillion GUIs out there (as
http://www.voip-info.org/wiki-Asterisk+GUI will attest).
However, I'm not sure which to use. A lot seem to be fairly
comprehensive... but until I kick the tires, it's trial-and-error.
And that would be a *lot* of trial-and-error.
Yes that is how AMP works. It's a very nice
setup.
- Original Message -
From:
Douglas
Garstang
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 22, 2005 5:07
PM
Subject: [Asterisk-Users] Creating conf
files from db
OK we need some help in setting up a good wiki-info page for setting up
the Mediatrix 1204 to work with asterisk. If anyone has set these unit's
up and have them working please post your settings here so we can create a page
on the wiki. These unit's are being sold to be used via sip format
[EMAIL PROTECTED] has the Zaptel from head. Just need to update the zaptel
drivers from CVS head you don't have to upgrade the asterisk.
Matthew Fredrickson wrote:
On Oct 27, 2005, at 12:02 AM, Mark Quitoriano wrote:
i tried doing the instruction from voip-info[1] anyway here's my
comment
Asterisk wrote:
Hello,
I am new to this list and to Asterisk. I am using Asterisk @Home, but
have begun to be comfortable editing the scripts.
I have a Grandstream GXP-2000 with 4 line buttons. Is there any way I
can set Asterisk to send more than one call to the phone without
setting up
Nathan Pralle wrote:
Hey folks,
Anyone know of companies selling bulk SIP adaptors (phones, adaptors,
etc.) or has the list ever considered doing something like a bulk buy?
Give a call to VoipSupply.com 800-398-VOIP (8647)
I was just curious...I'm looking to get another 5-6 Grandstreams or
They do not support faxing over there
network.
IP faxing has always been a problem. ulaw is the
only codec you can use to do this with. And for that matter of fact it's hit or
miss.
Ariel
- Original Message -
From:
Rene Nelson
To: asterisk-users@lists.digium.com
Alan Bunch wrote:
Ok, if I missed something in the wiki please point me there with the
correct search terms.
Asterisk 1.0.7 (AAH really)
4 co lines from Bellsouth into a Diguim T400P.
Polycom 501 x 4 on the desktops.
My problem is on calls to or from the CO I hear a beeping every 12
Colin Anderson wrote:
There is no real answer to your question.
just use one you're most familiar with.
I use RH allot so I am now using CentOS mostly. It's Red Hat EL GPL code and
so far everything I have runs on it without issues. Great OS.
Second that, using FC2 for me, and it's the
canuck15 wrote:
That is great news Ryan. I don't know what I would do without AMP.
One question. How can I upgrade from 1.10.008. I downloaded it and
followed the procedure in the UPGRADE file but it hung at:
Fri, 9 Sep 2005 16:28:55 -0700 - Unable to connect to manager
127.0.0.1:5038
Matt Fredrickson wrote:
On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote:
the echo isnt horrible most of the time, and seems extremely random
in that i can call a number once without echo, then dial the same
number a second time and get echo.
things i am currently considering (and
have you tried in the sip.conf for the
devices
canreinvite=yes
- Original Message -
From:
Tomas Florian
To: asterisk-users@lists.digium.com
Sent: Tuesday, August 30, 2005 8:48
PM
Subject: [Asterisk-Users] Registrar only
setup
Hello,
Im having
Sergio Serrano wrote:
If I use hearbeat I need a failover system for ISDN Lines, not? I
waould like that if Server A crashes, Server B Control SIP
Registration and ISDN Lines. Do you know about this?
There is a new product form Redfone that will help provide a failover with
the T1/PRI
jennyw wrote:
Dave Cotton wrote:
This was your first experience with *, was it also the installers?
Only sharing with the next busiest card in the machine the one
feeding the IP phones.
Yeah, I know, in retrospect it sounds really odd that we did that, but
at the time he thought there was
) For us to give you more help we are going to need to know more about you
system. What is the server your using? What phones? How is your network
setup? If you want you can email me directly. I will try to help you out
with your setup.
Ariel Batista
jennyw wrote:
Hi,
We recently tried
Tim P wrote:
[EMAIL PROTECTED]:mypass:[EMAIL PROTECTED]/2068660133
You need to add the number to the back so you can route it with asterisk.
Ok I can register with BV fine (as far as I can tell from asterisk -
see below). I am able to make outgoing calls but all incoming calls
get a fast
Chris Gamble wrote:
Just got in a bunch of polycom phones for use on my shiny new
asterisk box, but found 2 small issues I was wandering if someone
could help me with.
Are you using AMP or Asterisk @ Home?
First, though the phones support 2 call appearances, if I am on a
call, the second
This is simple since your using AMP,you can
create a ring group to dial that number out for you. First create your
ring group lets put number 200 for it (you can call it any number you want).
where the extension number goes just put there the phone number you want like
301212# don't for
I would need to see what the CLI displays
when this happens. I am using at many locations and it works. How many
dialing rules for trunks have you setup. Do you have any passwords
setup?
We have a user to user support area for amp on the
freenode #amportal You can find me there as well.
Adrien Laurent wrote:
Hello everyone,
I have an IAX server ([EMAIL PROTECTED]) with a FXO card.
I have a trunk connected to a voip provide, asteriskout.
When I call my server from a public phone, I want to route this call
to the asteriskOUT trunk so that I can make long distance calls.
Your
C F wrote:
The IP-501 AFAIK comes shipped with SIP1.5.2 which does NOT use
ipmid.cfg. You have to get new *.cfg files for the ip-501 or the older
phones that run 1.5.2.
Sorry to tell you but that is not a correct. The IP-501 I have about 15 of
them new and they came with 1.4.2 also they do
]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel
Batista
Sent: Friday, 15 July 2005 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Gui?
Michael Felder wrote:
Can anybody recommend an Asterisk GUI to help a newbie confg ?
Try [EMAIL
Michael Felder wrote:
Can anybody recommend an Asterisk GUI to help a newbie confg ?
Try [EMAIL PROTECTED] it's a complete ISO with everything you need to start
with.
If you already have the OS installed then check out AMP which is by the way
included in [EMAIL PROTECTED]
Kind regards
Sahil Gupta wrote:
Hi,
Is there anybody on the list that recommends anyone for
colocation/telehousing in the US?
I'm after 2 Servers to be hosted in the US, preferably on the west
coast.
I would suggest www.race.com
Regards,
Sahil Gupta
VoiceValley
Wow,
I just want you to know I am and have been a
Networks Engineer for many years. I started back when Novell was king for
networks. Window and many others have come by and I have setup shop with
them. I still manager and maintain several of my Clients Windows
networks. Almost 3 years
Brian Roy wrote:
On 6/15/05, Rich Adamson [EMAIL PROTECTED] wrote:
In other words, the further the spa3000 (or TDM card) is from the
central office, the more difficult it seems to be to set gain values
that are acceptable. That's apparently why many people find its use
is okay while others
Title: MCI vs. XO/Allegiance
we have been using XO/Allegiance for over 3 years
and have had no problems. I can't compare to MCI but we also had a sprint
t1 that we had to get remove due to them being bad in billing and also not very
reliable for faxing.
- Original Message -
Justin Ellison wrote:
Hey all,
Just getting started playing around with my Polycom 600. According to
the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP
1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1
and 1.5.2?
I am still running BootRom 2.6.1 with
[EMAIL PROTECTED] wrote:
Hi,
I recently got a TDM04B and after installing and getting asterisk up
and running I connected a handset to one of the ports. Unfortunately
I don't get a dial tone when I lift the handset.
This board is FXO which you plug incoming phone lines into it. So plugging
I have looked at the wiki and the mailing list. But
I need to find how do we setup the external IP address and the rtp ports for the
Polycom IP-500 and IP-600. There web interface has a nat setting but can't
find instructions on how to set this up. I would like to set this up via there
ftp
Asterisk @ Home This CD will install
everything you need to get your Software PBX going.
Its a complete ISO CD that brings
together the OS (CentOS 3.4)
Asterisk Software version 1.0.7 stable
AMP Asterisk Management Portal
Web GUI
FTP
TFTP
Plus many more items.
Every
Sean,
We setup a support department via just that way. In fact it's about the only
real way to get modems working correctly. We used T100p card attached to
Adtran 750 units. We got them on ebay for around $ 500.00 each. Which is
well worth the cost.
Good luck.
-Original Message-
From:
They work just fine. I have a few of them out at customers and there working
without any issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Geoff Manning
Sent: Friday, May 20, 2005 3:01 PM
To: Asterisk Users (E-mail)
Subject: [Asterisk-Users] Dell
Well, if you are only making office to office calls, save the $500 per
T1 card and just use NICs.
The T1 card is only required if you are using a voice T1. If you are
doing IAX to IAX for example between offices, then Asterisk is your
friend.
Avoid SIP altogether as it is not needed and
I know that you can contact www.race.com they have rolls of rack space
available in One Wilshire in LA.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of InternetMarketingMan2001
Sent: Friday, May 20, 2005 6:25 PM
To:
asterisk-users@lists.digium.com
Subject:
Just want to let everyone know that even if there changing it out to the new
501 it's still on of the best. Remember that people are still buying the
Cisco 7960G which is being phased out as well.
The IP-500 works and works very well. I know that there price will be going
down soon once there are
I have 2 of them working on a SC420 server and also another one the SC400
and older one that has 4 TDM boards on it. Both systems have been working
fine.
I did not have to do anything special on them to get them working.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hold on asterisk is not possible But you can park the call. If you have
setup parking extensions in the features.conf the default is 700. You would
flash then 700 it will give you an area where it parked like 701. Then when
your ready to get back to the call just pickup and dial exten 701 and you
I would say it would be batter to the the TDM11b since it will have your
inbound analog line for 911 and faxes and the FXS port you can plug your fax
machine in. This is what I do for most of my SoHo setups. Which I also use
AAH for.
Ariel
-Original Message-
From: [EMAIL PROTECTED]
Yes I have customers using this switch and
the 2324 as well.
They work fine even with the IP-500
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera
Sent: Wednesday, May 18, 2005 4:51
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
You need to go into the extensions setup and put the pickupgroup and
callgroup to the same on both. That way when you hear the other extension
ring you just dial *8 send and you can pickup the ringing phone call.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
The receptionist phone is going to be a hard one. We use Flash Operator
Panel. Works great.
Now about the phones for all around great phone we are using the Polycom
IP-500 which is in my view one of the top of the line phones.
For el cheapo well we are using one that is yes cheap but also pretty
I don't have any problems using a pots line with the credit cards. In fact I
have in some locations a Sipura that is attached to the cc machine. Works
just set it up just like a fax using ulaw only.
Ariel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have gotten Digium boards, Sipura and Polycom phones from them. There very
good and I have not had any problems with them.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Manjit Riat
Sent: Tuesday, May 17, 2005 8:24 PM
To: 'Asterisk
Check features.conf for parking extensions. There default is 700
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
Sent: Monday, May 16, 2005 8:51 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Help
It would be nice if you post how you set this up to either the wiki or right
here. Just a few lines would do nicely. There seems to be allot of people
who use voipjet and aah and both are good products.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I just setup one Dell SC420 with just one SATA drive and 512mg Ram ($
404.00) with 2 TDM04B in it that is 8 FXO ports. And a second system for
another customer with 3 TDM 2 TDM40B 8 FXS ports and one TDM01B for 4 FXO's.
Both systems are working just fine.
-Original Message-
From: [EMAIL
Yes it's used in Ft. Lauderdale.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo
Sent: Wednesday, May 04, 2005 10:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] 10 digit dialing in Ft Lauderdale, FL?
Does anyone know if 10
I just setup a SC420 with two TDMO4b cards
in it and it works just fine. No problems what so ever with it so far.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Hencke
Sent: Friday, April 22, 2005 6:42
PM
To: Asterisk
Users Mailing List - Non-Commercial
If your used to RH keep using it. Since I am a person that has used RH for
many years I have gone with CentOS which is RHEL via GPL. It's great and
there yum servers are always up and running.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Shiflet
From what I have heard it works but has still some issues.
It's on sale from VoipSupply for 114.95
http://www.voipsupply.com/product_info.php?cPath=95_111products_id=331
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Normandin
Sent: Wednesday,
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio
Sent: Monday, April 18, 2005 9:16
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] queue -
transfer calls
Hello,
I am setting up an ACD using *, but found a an issue
If there is a mistake that you can fix then do so. If it's your option to do
it differently then add a note to it and put your text after that. The Wiki
is an open and should always be open. But as you stated Etiquette goes 2
ways and you should not go around removing text because you don't
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy
Sent: Monday, April 04, 2005 6:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Channel bank question
Hi all,
Quick question regarding channel
Welcome,
Yes I have used it. It's great to get started. Give it a try.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bagan Jermal
Sent: Friday, March 25, 2005 6:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hello Everyone
would
Hello I am looking for 2 things to add to our Asterisk
servers.
I would like to know if there is any way to monitor the
PRI/T1 lines via the Asterisk Server to see if they go down. If they go down
then email a notice to us. Also would like to extend this to if the Asterisk
goes down
We all at one time were in the same boat. Well my suggestion is do allot of
reading on the wiki for asterisk, 2nd get your self a PC load [EMAIL PROTECTED]
and start working with it. It has everything you need to get started. And
it works. Now you don't have to keep it with the supplied .conf
All the samples are on your system
/usr/src/asterisk/configs/ the files have a .sample on them.
Also there is allow of information on the
Wiki http://www.voip-info.org/wiki-Asterisk
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bharat M. Sarvan
Sent:
This question really has no one reply. The different Linux builds all have
there reasons. If your used to Fedora Core 1 then that is what you should
use. I use CentOS which is a clone of RHEL 3. They have just released there
Version 4 which is based on RHEL 4. It works and since I am used to
You need to call PBXware it should not have anything to do with the phone.
If a phone registers there gui should put it in the correct context. If it
does not then get your money back due to there non standard setup
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Try this site: http://fedoralegacy.org/ they have most of the things there
for RedHat 7.1 on to Fedora Core 1 items.
- Original Message -
From: Muhammad Muzzamil Luqman [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, February 18, 2005 1:48 AM
Subject:
Great setup for the [EMAIL PROTECTED] .06.
I have a few questions about the console mode. If you go to the Ctrl Alt F9
area you see asterisk loaded but it's displaying some funny Greek letters. I
did the following but it did not help.
Eliminating some internationalization errors:
In
Yes turn off silence suppression.
xlite - Menu - Advanced - audio settings - Silence
Settings - transmite Silence: (change to yes)
- Original Message -
From:
Julius
Kidubuka
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 10:04
AM
files in /etc/init.d I think only a clean
install will fix this.
3. A lot of changes in FOP too the config files are in
a different place could cause this problem.
Sorry about all the changes. As we get closer to a 1.0
release of [EMAIL PROTECTED] a lot of this will stabilize.
--- Ariel Batista
Roger Hanson wrote:
I've downloaded 2x and burned 2 cds and get an error invalid
compressed format (err=2) system halted message both times.
It'd be nice to have a MD5 to verify my download is OK. It'd narrow
down the problem to either the download or the burn, wouldn't it?
The other day I was
[EMAIL PROTECTED] wrote:
We are releasing a new version of our one-button
Asterisk install, [EMAIL PROTECTED], today. This release
includes a redesigned web interface and auto-detection
of Digium fxo and fxs cards. We have also fixed a lot
of bugs and added numerous customer requested
Hello,
Great job on the [EMAIL PROTECTED] project. Looks great this new
version is really nicer looking. But I have a few questions.
1) For the new web access http://localIP/mainthow and where do I
change the password.
2) Since I don't use the Amp section for setup the
conf files I use my
Thank you that worked for the passwords..
Thanks
- Original Message -
From:
dean
collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, February 11, 2005 10:06
PM
Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions
Asterisk wrote:
I have managed to get spandsp working, and if I dial a specific
extension I can receive faxes. WhooHoo.
However, I was wanting to use the fax detect option in order to
allow individuals to receive faxes, but can't get that to work.
Given the following extensions (mainly copied from
Title: AMP with SUSE9.2 (Apache2)
I think you should post this information on the
Wiki. It's our main location for all of these things.
Thank you.
- Original Message -
From:
Keith
Burns
To: asterisk-users@lists.digium.com
Sent: Thursday, February 03, 2005 2:09
Well I just need to say I got my phone last week.
Here is my quick review of the phone and hope that someone has a possible fix
for it or I will be sending it back.
First the phone is nice looking in my view and it's
heavy so it feels like a real desk phone. But it has these stick, gummy or
I just have to make my view known about this.
1) I agree that one is needed but!
2) I feel that there should be a way to get a self
study course which will lead to a way to take a test for the
Certification.
3) Cost and who set this up is really something
that I think should be done first
Eric Hall wrote:
I installed spandsp on our asterisk server to get faxes. It works
however the images are a little off. Sometimes a few pages will be
together, pages missing and sentence missing.
Is this normal for this program?
Yes it is with some fax machines. We had to make our own program
Warren Burstein wrote:
One more thing about prompts, it's better to say for sales press 5
than press 5 for sales, because by the time you hear sales you've
already forgotten what number it was.
If you add the sounds all you need is For Sales recorded the new sounds have
press # already. So you
Original Message
From: Ariel Batista
I have Spandsp working fine. Asterisk sees a fax on the zap port and
redirects the call to the fax-in area.
This works if I have a simple dialing rules that goes answers first
and waits 10 secs then goes to the next item. If it hears a fax it
goes
I have Spandsp working fine. Asterisk sees a fax on
the zap port and redirects the call to the fax-in area.
This works if I have a simple dialing rules that
goes answers first and waits 10 secs then goes to the next item. If it hears a
fax it goes to the right place. Here is a sample that
a better voicemail program than this one.
-
\
\\_ Ariel Batista
//
/ Avionica, Inc.
--
[EMAIL PROTECTED]
Ph: 786-544-1114
Fx: 305-574
John Todd wrote:
At 12:00 PM -0400 on 7/23/04, mattf wrote:
Hello,
I recently tried an upgrade of CVS on my test server today and found
that the res/res_parking.c file is completely gone. This is where I
had to go into the code every time I do an upgrade and change the
code to allow for
We use Allegiance it's been very good to us. It's
now X/O but still we have had no outage. We have a ATT and a Sprint
line as well. Sprint is evil and sucks. ATT is good but for the price I can
get more from Allegiance down here in South Florida.
- Original Message -
From:
Arick Davis wrote:
www.Kall8.com
Expensive .068 that is 6.8 cents per minite.
And they support SIP termination.
Arick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Laird Sent: Monday, June 14, 2004 9:29 AM
To: [EMAIL PROTECTED]
but
outside address section?
-
\
\\_ Ariel Batista
//
/ Avionica, Inc.
--
[EMAIL PROTECTED]
Ph: 786-544-1114
Fx: 305-574-0212
Kubat, Philip wrote:
I have an old Dialogic D/41E card. I searched the mailing list and
it looks like there was or could be a module for it. Although the
posts never specified where or how.
Is a D/41E usable w/Asterisk? If so how does one obtain the drivers?
Or is it a better pots
Timothy R. McKee wrote:
My SIP users need to transmit the # key as part of data entry.
Asterisk intercepts and initates a transfer function. I'm almost
positive I've seen this discussed somewhere, but none of my searches
are finding it.
In your dial plan take the Tt out of it. exten =
is extermly large so it is not a good idea to post it here.
-
\
\\_ Ariel Batista
//
/ Avionica, Inc.
--
[EMAIL PROTECTED]
Ph: 786-544
working fine they all can park a call and pick them up.
-
\
\\_ Ariel Batista
//
/ Avionica, Inc.
--
[EMAIL PROTECTED]
Ph: 786-544
Osvaldo Mundim wrote:
Hi,
Does anybody have ATA 188 working with any kind of fax machine? I've
tried many different configuration following the Cisco Online Manual
and I couldn't get this working with Asterisk.
I don't know what the difference is between the 186 and 188 other then the
extra
then need to reboot the system.
-
\
\\_ Ariel Batista
//
/ Avionica, Inc.
--
[EMAIL PROTECTED]
Ph: 786-544-1114
Fx: 305-574-0212
James Moran wrote:
Anyone have any suggestions on free sip phone software for windows??
Only have one IP phone and want to have one other computer hooked up
to my Asterisk box for testing.
xten x-lite. Works great free and works just plain works!
Mark Spencer wrote:
Any feedback on:
a) The idea itself -- is it a good one or is it stupid?
Now this is just my views. No I do not feel we need to be sending any
information back unless we want to. Like someone else said a sub job that is
turned off by default. My preference would be no
Bisker, Scott (7805) wrote:
I've been trying to get a Win 2000 RAS server working with my
asterisk PBX for quite some time, to no avail. I've googled, I've
tried loads of configurations, I've rewired phone lines, and still I
am not winning the battle.
Here's my config.
Bisker, Scott (7805) wrote:
Same as mine. Do you know off the top of your head what firwmare
you're using? Also, what RAS card do you have on your PCAnywhere
side?
I have firmware L36. Ras card is a Digikey 4 port board on one NT server
and others are using the normal serial ports on the
might have the same problem with there lines like
Sprint. That there not data lines but voice only.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ariel
Batista
Sent: Wednesday, April 07, 2004 12:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk
Christopher C. Howard wrote:
Andrey McRory built a RPM dist for * but I can't seem to find it
anywhere.. Any hints where I might be able to find this package that
has matching kernel?
This is what I found for rpm. http://www.voip-info.org/wiki-Asterisk+RPM
Hope this helps.
Thanks,
Chris
Gary Franczyk wrote:
Hello,
We are trying to deploy a new asterisk server with a Wildcard T400P
(quad T1) card. It uses a custom voice recording app written in the
perl AGI.
Now that the machine has been in production, it seems to lock up
within 24 hours of reboot! When it locks, we can
1 - 100 of 178 matches
Mail list logo
