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maxsilence in voicemail.conf
- Original Message -
From: Robson Ribeiro [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 6:21 AM
Subject: [Asterisk-Users] TDM04B doesn't hang up after Voicemail
Hello all,
I am having a serious problem installing my *
here is what i get
speex -11 5 511 5 412 - -43
- Original Message -
From: Kerry Garrison [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 10:13 AM
[EMAIL PROTECTED]
- Original Message -
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 12:51 PM
Subject: RE: [Asterisk-Users] AMP-1.10.007 Released!
Ryan,
I had this problem in Africa. I was not provided any docs fromt he telco
but the problem was the crc4. when omitted everthing was fine.
- Original Message -
From: Henry Jensen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 12:04 PM
Subject:
Title: a simple question .
save this http://www.szmidt.org/asterisk/asterisk-update.shto
/usr/src and run it.
you can also go into /usr/src/asterisk and type
make update
- Original Message -
From:
Weiming Jiang
To: asterisk-users@lists.digium.com
Sent: Friday,
Hello everyone:
I've had X100P running on asterisk 1.0.6 for about two
months. Each time I start linux, I manually modprobe
zaptel, wcfxo, and ztcfg before I start asterisk.
Today, all of the sudden I am not able to load ztcfg
and I get the error message
FATAL: Module ztcfg not
1/3 down the page are your answers.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
- Original Message -
From: Norman Zhang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 28, 2004 7:20
IAX2
- Original Message -
From: Serge Schumacher [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 31, 2004 8:00 AM
Subject: [Asterisk-Users] IAX users
Hi,
I do not understand the difference
you could also forward port 4569 from the nat router to your asterisk box.
i think a qualify statement in iax.conf will also help if you find out how
quickly the router is shutting down the map and set the qualify statement to
a shorter time frame.
- Original Message -
From: Dinesh Nair
On Mon, 10 Jan 2005 20:09:54 -0600, Rich Adamson [EMAIL PROTECTED]
wrote:
Does asterisk support the intel 537/md3200 chipset? I don't want to
start
any flames here, I know all about using generic crap in asterisk,[*]
which
I really don't approve of other than for testing, but I have
you have a document with
descriptions?
thanks
rafael
On Sat, 4 Dec 2004 10:06:34 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
Same thing here. It used to work perfectly until I re-installed.
Hi
I need some advice in this issue, I installed astcc again and creates
database from configure
with
descriptions?
thanks
rafael
On Sat, 4 Dec 2004 10:06:34 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
Same thing here. It used to work perfectly until I re-installed.
Hi
I need some advice in this issue, I installed astcc again and creates
database from configure menu but I am still getting errors
Same problem. I dropped all the tables and ran the above. I also checked
the sockets and they are correct. Any other ideas?
SNIP
I am still stuck on this.
/SNIP
Allright, let us do it the easy way. Just copy and run the script below
on your
Mysql Command line. Lets see how this goes.
If you
Steve,
What version of MySQL are you running? I upgraded to 4.1.8 and ran into
the
problem below. I initially tested with the user root and the default
blank
password and was OK. But when I changed over to a new user with a
password,
I noticed an error message in the httpd logs:
Steve,
What version of MySQL are you running? I upgraded to 4.1.8 and ran into
the
problem below. I initially tested with the user root and the default
blank
password and was OK. But when I changed over to a new user with a
password,
I noticed an error message in the httpd logs:
If nothing else, my efforts are documented for anyone else in the
same boat.
It seems that you can debug agi by typing agi debug at the * command
line.
Amazing! Here is the output. I am assuming that since astcc-tone didnt
play, the problem lies there. Thoughts?
Steve,
I just
If nothing else, my efforts are documented for anyone else in the
same boat.
It seems that you can debug agi by typing agi debug at the * command
line.
Amazing! Here is the output. I am assuming that since astcc-tone
didnt
play, the problem lies there. Thoughts?
Steve,
You can change the setting. I set mine for
every 1 min on a small system. The phones always work.
What is the register
interval in the grandstreams? The qualify=yes should keep the connection alive
as long as Asterisk is up, but if it goes down and then comes back up, the
phone
If nothing else, my efforts are documented for anyone else in the
same boat.
It seems that you can debug agi by typing agi debug at the *
command
line.
Amazing! Here is the output. I am assuming that since astcc-tone
didnt
play, the problem lies there. Thoughts?
There is a mailing list dedicated to AMP that would be better suited to your
line of questions.
To answer your question, you can create your own files and name them
whatever you want and then use include statements. The most you will have
to re-input are the include statements.
- Original
Paul Fielding wrote:
- Original Message - From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grandstream Bugetone 101 mwi
I have a notebook setup with AMP and WBEL about ten remote phones and IAX
connectivity to a provider. It is a complete demo system with a DID pointed
to it. I can either take a phone and plug it in at the customer site and
make and receive calls or even take the laptop. Make sure you setup
You can have voicemail disconnect after a
predetermined amount of silence in voicemail.conf.
maxsilence=5
silencethreshold=128
- Original Message -
From:
Ken
Knight
To: asterisk
Sent: Thursday, January 13, 2005 1:11
PM
Subject: [Asterisk-Users] asterisk won't
We have one Bt-100 that logs on to the server,works for a few min and
then just starts loosing registration
There is one recent firmware version that has a registration fallout
problem. I'm afraid I don't know which version. I have my own problem
I am discussing with GS.
Any firmware
You shouldn't need any port forwarding. I've found any SIP phones I've
worked with have happily moved from site to site behind NATs, etc. So I
see
no reason to believe that the WiFi phone would be any different - it's
just
connecting wirelessly instead of with a wire.
And to answer
try iax2 debug
Hello,
I've signed up for a NuFone account, and added the following
instructions to my config files per NufFones directinos:
iax.conf
[NuFone]
type=peer
host=switch-1.nufone.net
secret=password
extensions.conf
(under the [default] context)
exten =
Are you almost done sorting the files?
- Original Message -
From: Rob Fugina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; Asterisk Developers Mailing List
asterisk-dev@lists.digium.com
Sent: Thursday, January 13, 2005
I have a need for a system that will strictly be used for authentication and
minutes used when a caller dials in. The system needs to support 8
incomming lines and 8 outgoing lines into a proprietary call center box and
PBX (3Com NBX)
What is the best way to accommodate this with *? I was
I have a need for a system that will strictly be used for authentication
and
minutes used when a caller dials in. The system needs to support 8
incomming lines and 8 outgoing lines into a proprietary call center box
and
PBX (3Com NBX)
What is the best way to accommodate this with *? I
does anyone know how to change the timeout on digit entry in astcc. if you
call the app and start entering a pin, you have about 2 seconds to enter the
next number or you get timed out. i cannot find any info on this from the
lists or google.
Thanks,
Steve
it reviews the Link, can help you
http://asterisk.drunkcoder.com/agi.cgi
greetings
Rodrigo
El Jue 03 Feb 2005 14:02, Steve Totaro escribió:
does anyone know how to change the timeout on digit entry in astcc. if
you
call the app and start entering a pin, you have about 2 seconds to enter
the next
AMP is great and provides integrated extensions to dial to access features
implemented in asterisk. It also provides webbased access to voicemail and
flash panel operator. You will find that many phones have features
implemented into them as well. The snom phones are a great example of this
and
not a permanent solution according to many on the list but try type=friend
in your iax.conf
- Original Message -
From: Sergey Kuznetsov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 3:40
I have been playing with a Clipcom that is pretty cool.
- Original Message -
From: Olaf Klein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 16, 2005 2:03 PM
Subject: [Asterisk-Users] WLAN-Voip phones anyone?
Hello,
Does anyone here use any WLAN
I would eliminate everything that is not necessary,
like the amaflags and the auth=, account code stuff. I would also use IP
address rather than domain and get it working. Then I would start adding
the extras back in.
- Original Message -
From:
Sergey Kuznetsov
To:
They did make a SIP phone and are about to release new SIP phones and a new
product line. The old SIP phones look identical to the NBX phones but I am
not sure about the guts. Possibly the 2102 could be flashed into a 1002.
Here is an ebay auction but these phones are really hard to come by. I
Search on DTMF
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 17, 2004 2:57 PM
Subject: Re: [Asterisk-Users] can't logon to voice mail - bad password
Paul,
Do your other extensions work?
If you have only one extension, note that the filename
that is SWEET!!!
- Original Message -
From: Philipp von Klitzing [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 9:02 AM
Subject: Re: [Asterisk-Users] Re: Grandstream 1.0.5.0 Firmware: SIP Register
option gone
Hi!
Did you try out the new ring tones? One of
Can we see your extensions.conf
- Original Message -
From: John Campbell [EMAIL PROTECTED]
To: asterisk-users [EMAIL PROTECTED]
Sent: Tuesday, June 08, 2004 3:55 PM
Subject: Re: [Asterisk-Users] Don't want a ring before voice menu
I should have been clearer in my description of the
Is anyone else having problems right now.
Only about half the times that I call my DID does it go through. I am not
getting a fast busy either, I get dead air.
When the call does go through it is VERY
choppy.
Thanks,
Steve Totaro
www.totarotechnologies.com
I like the way the 3com NBX system works. The web interface is pretty
intuitive. Adding users and devices is a snap through the GUI but to get to
the real meat you have to edit the dial plan. To do this, you download a
text file to your desktop, edit it, then upload it again.
- Original
I was having the same problem. Only about half the incoming calls were
getting through, then completely down. Now it seems to be back up.
While it was down I ugraded to today's head and it started working again.
The two are probably unrelated.
- Original Message -
From: Wojciech Tryc
I have tried it with 4 simultaneous calls and it worked like a charm.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 12, 2004 8:41 AM
Subject: Re: [Asterisk-Users] 2 NuFone lines- which one to dial out on
On Sat, 12 Jun 2004, Simon Dorfman
goes for Voicepulse. I've
been using
it for multiple inbound and outbound calls for about a month.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Saturday, June 12, 2004 12:44 PM
To: [EMAIL PROTECTED]
Subject: Re
Title: Message
I think he just wants to promote
gafachi.com
- Original Message -
From:
Jay Milk
To: [EMAIL PROTECTED]
Sent: Monday, June 14, 2004 11:59
AM
Subject: RE: [Asterisk-Users] making *
more like a normal pbx
You
really need to start making
Sounds like good questions to ask VoicePulse.
- Original Message -
From: James W. Brinkerhoff [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 14, 2004 1:14 PM
Subject: [Asterisk-Users] Number Portability and VoicePulse
I have two questions regarding number portability...
[EMAIL PROTECTED]On Jun 14,
2004, at 10:48 AM, Steve Totaro wrote:
I think he just wants to
promote gafachi.com-
Original Message -From:
Jay Milk
To:
[EMAIL PROTECTED]
Sent:
Monday, June 14, 2004 11:59 AMSubject:
RE: [Asterisk-Users] making * more like
I would like to setup a system with two
tennantswith twoseperate DIDs through one IAX provider
account.
Is it possible to route the calls into different
contexts based on the DID dialed?
I have searched and found nothing. I do not
see anywhere in the console that says what DID was dialed
LIFO Last In First Out
- Original Message -
From: tmpm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 4:30 AM
Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette
Its a matter of personal preference Holger, most people dont care, but the
ones who do
I hate asci signatures that are hard to read, stop eating up my bits :p
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 6:07 AM
Subject: RE: [Asterisk-Users] Asterisk-Users List Etiquette
Steven Critchfield [EMAIL
I can ping it just fine.
I am on
gw5.voicepulse.com
Down Again?
This is not the place for these typs of
messages.
bkw
- Original Message -
From:
Steve Totaro
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 6:50
PM
Subject: [Asterisk-Users] Voicepulse
Down Again?
I can ping it just fine.
Just curious if you are actual list moderators or self-appointed moderators.
- Original Message -
From: twisted [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 9:46 PM
Subject: Re: [Asterisk-Users] Voicepulse Down Again?
On Tue, 2004-06-15 at 19:20, Steve Totaro
PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 9:46 PM
Subject: Re: [Asterisk-Users] Voicepulse Down Again?
On Tue, 2004-06-15 at 19:20, Steve Totaro wrote:
Then where is? A good many Asterisk Users use voicepulse connect so
I would say it does.
sgt
- Original Message
I only have experience with the first problem. This happend to me using
netscape on a linux box but has never happend using IE on a windows machine.
- Original Message -
From: Simon [EMAIL PROTECTED]
To: Asterisk-Users [EMAIL PROTECTED]
Sent: Wednesday, June 16, 2004 9:14 AM
Subject:
iax to
Voicepulse Connect. Are you aware of any problems? I have published this
number to customers.
Thanks,
Steve Totaro
Totaro Technologies, Inc.
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 15, 2004 10:05 PM
Subject: Re
I think it should be without the quotes like this:
exten = 555,2,Festival(good morning)
- Original Message -
From: Freddy Setiawan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 1:50 AM
Subject: [Asterisk-Users] Festival and asterisk
I've install the asterisk in
I would also like to know how to insert a pause if possible. A comma is
seen as | not surprisingly.
I have no idea why no quotes.
- Original Message -
From: S. William Schulz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 19, 2004 10:24 PM
Subject: Re: [Asterisk-Users]
)/Are_you_looking_for_an_affordable_way_to_add_VoIP_capabilities_to_your_existing_PBX_phone_system?/28/
Thanks,
Steve Totaro
Totaro Technologies, Inc.
410.985.7114
- Original Message -
From: Florin Andrei [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 21, 2004 3:19 PM
Subject: [Asterisk-Users] integrating with existing PBX
I'm looking
Any easy way to make festival read the name?
- Original Message -
From: Harold Workman [EMAIL PROTECTED]
To: 'Bruce Komito' [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Monday, June 21, 2004 1:24 PM
Subject: RE: [Asterisk-Users] Directory dial by name
Bruce Komito wrote:
Directory only
Grandsteams can be configured to dial an extension as soon as you pick it
up. Great for doorphones and your application.
I forget the exact vebiage but I know its in there. Oh here it is, Offhook
Auto-Dial:
- Original Message -
From: PAZ [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
I am trying the same thing. Now I get this message in /var/log/messages
after trying modprobe zaptel.
kernel: zaptel: version magic '2.6.5-1.358custom 686 REGPARM 4KSTACKS
gcc-3.3' should be '2.6.5-1.358 686 REGPARM 4KSTACKS gcc-3.3'
This is fedora 2 core. Any ideas?
Thanks,
Steve
-
apologies for sending that attachment to the
list
- Original Message -
From:
San Singhania
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 9:43
AM
Subject: [Asterisk-Users] Asterisk
Article
Hello,
sometime back, I saw an article (i think it
It sounds like you probably had a fractional t-1 with 3 DIDs (probably more
that you didnt use). Did your data also go through this pipe? Get a copy
of your bill from the phone company.
Get a decent server. The beefier and more redundant the better. Get a
single span t-1 card. As soon as you
Let them suffer like the record industry. Times are changing, better change
with them or fall by the wayside.
- Original Message -
From: Joe Baptista [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 06, 2004 5:19 PM
Subject: [Asterisk-Users] VoIP under attack ... Bellcos
Are there any sites setup to let the powers that be, know that some (voters)
people are against these kind of moves? If not, lets get it going.
arguing the homeland security aspect may be difficult in the current
environment but FEES can only hurt an emerging technology!
- Original Message
I had this problem when I split the line to a dialup modem before it went
into the * box. Everytime a call was placed by the modem, * saw it as an
incoming call. This was a while ago though.
Steve Totaro
- Original Message -
From: Jason Williams [EMAIL PROTECTED]
To: [EMAIL PROTECTED
why regulate? nobody regulates the return address on a letter sent via
USPS.
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 10:00 AM
Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
Adam Hart [EMAIL PROTECTED]
qualify
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20qualify
- Original Message -
From:
Chris Smales - Magenta
Solutions
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 10:09
AM
Subject: [Asterisk-Users] UDP Ports scan
on firewall
I liked the NuFone chief Jeremy McNamara didn't return phone calls for this
story.line. ;-)
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 11:45 AM
Subject: Re: [Asterisk-Users] VoIP hackers gut Caller ID
Chris
use the wiki as a reference
www.voip-info.org
- Original Message -
From: kaiduan xie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 12:10 PM
Subject: RE: [Asterisk-Users] Newbie's doubt on sip.conf
Andrew,
Thanks for help. But you
The power cord it for fxs ring voltage I believe. Shouldnt be needed for
fxo
- Original Message -
From: Gelson Dias Santos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 1:24 PM
Subject: Re: [Asterisk-Users] Unreliable dtmf digit generation from tdm400p
I
yeah from the same guy that reprimanded me for asking a question that he
deemed not appropriate for the list. i asked if voice pulse connect was
down.
- Original Message -
From: Kevin Walsh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 2:12 PM
Subject: RE:
to
technology). They have been merging messages into images and posting them
on the internet for years. That takes more know how than placing a voip
call.
Thanks,
Steve Totaro
- Original Message -
From: Brian Cuthie [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 08, 2004
They could spoof their caller ID to the Bush/Cheney campain and call people
at 3AM to ask for their support!!!
- Original Message -
From: John Fraizer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 07, 2004 7:56 PM
Subject: Re: [Asterisk-Users] Kerry/Edwards campaign and
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
the wiki seems to be VERY complete when it comes to GS
- Original Message -
From: Bruce Komito [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 9:31 AM
Subject: [Asterisk-Users] sample config file for
the
manufacturer,
surely we shouldnt be reverse engineering?
Steve
On Thu, 8 Jul 2004, Steve Totaro wrote:
http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone
the wiki seems to be VERY complete when it comes to GS
- Original Message -
From: Bruce Komito
to write a book on Asterisk, which has been long
overdue.
DH
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Totaro
Sent: Tuesday, June 08, 2004 8:16 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FINALLY! a good book about
I built a system and then changed the IP and
subnet. Now the phones will not register, getting a 403.
Any ideas?
yes
- Original Message -
From: Ethan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 13, 2004 3:22 PM
Subject: [Asterisk-Users] Rotary phones? (No, I'm serious)
Will the FXS cards that work with asterisk handle rotary? Are there any
channel banks that can convert rotary
LOL,
IMHO=In my humble opinion
LOL=Laughing out loud
- Original Message -
From: ruixun wu [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 10:07 AM
Subject: Re: [Asterisk-Users] How to uninstall Asterisk?
hi Gus and Roger,
Thanks for you reply. I choose no
solution please.
- Original Message -
From: Christopher L. Wade [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 16, 2004 7:47 PM
Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Please disregard, I have 'solved' the issue.
Thank you,
Chris
they all use rj11. regular phone line cord.
- Original Message -
From: Florin Andrei [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 16, 2004 6:11 PM
Subject: [Asterisk-Users] PSTN/phone/FXO/FXS cabling issue
I just received a Wildcard TDM400P by FedEx yesterday. I
it can also be defined on some devices like the grandstreams.
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, July 18, 2004 9:42 AM
Subject: Re: [Asterisk-Users] Hotline
On Sunday 18 July 2004 09:36, Junaid Uppal wrote:
I tried
same problem here. the display shows vulcan.
- Original Message -
From: Mark Elkins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Monday, July 19, 2004 7:39 AM
Subject: Re: [Asterisk-Users] Brain-dead Grandstream BT102?
On Sun, 2004-07-18 at 23:52, Bruce Komito
I am not having much luck searching on why the
transfer button on my grandstream bt102 stopped working. anyone have any
ideas where to look?
i like the pingtel phones.
www.pingtel.com
- Original Message -
From:
Surajee
Ratnayake
To: [EMAIL PROTECTED]
Sent: Saturday, September 06, 2003 4:40
AM
Subject: [Asterisk-Users] What is the
best IP phone?
hi,
Can anybody suggest me a good,
for me but I wonder if there is a bad
batch of cards? I was in the "backordered till Sept 2nd batch" I am
assuming its not good to fail calibration?
Steve Totaro
- Original Message -
From:
How Peng
Kaiam
To: [EMAIL PROTECTED]
Sent: Thursday, September 11, 2003 9:52
www.sipphone.com
- Original Message -
From: John Brown [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 15, 2003 4:23 PM
Subject: Re: [Asterisk-Users] Grandstream Source?
Yes, you can now purchase GS phones from us
http://www.chagres.net/products/voip/phones.html
It seems we have a mailing list:
THE NATURAL LIFE CYCLE OF MAILING LISTS
Every list seems to go through the same cycle:
1. Initial enthusiasm (people introduce themselves, and gush a lot about how
wonderful it is to find kindred souls).
2. Evangelism (people moan about how few folks are
then ignore the thread.
to use your words...I always amazes me how so many EXPECT so much for
nothing...
- Original Message -
From: John Vozza [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18, 2003 1:19 PM
Subject: Re: [Asterisk-Users] Grandstream Source?
BS! :)
Look at all the time you are wasting flaming people. just ignore these
questions and get off the high horse. Do you maintain this list? If not
then you have no say whatsoever.
- Original Message -
From: Steve Creel [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, September 18,
i am just curious how many * systems are in the
real world with more than one user. do you run a certain version?
you dont update CVS do you? any admins running a system of over
twenty? over fifty? over one-hundred?
i deal with 3com and nec systems all day (i am
cerified in the 3com nbx
Can you disable your firewall? i am about to start this phase of asterisk
an would like help from one newbie to another. otherwise this newbie will
let you know how i did it.
- Original Message -
From: Brad Waite [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, September 20,
I would love to have a client that has _less_ of these advanced features :-)
I
will supply my users with a binary package and a registration file that will
configure the client to work with my server. After that, all references to
different servers would be omitted, and made invisible. No more
i am excited too.
what kind of wireless wan?
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, September 20, 2003 11:39 AM
Subject: Re: [Asterisk-Users] how many production systems are there?
Steve Totaro wrote:
i am just curious how
http://www.multitech.com/PRODUCTS/Families/SocketSLIC/
I am getting the following error in Windows Media
Player Version 9 when listening to voice mails.
ClassFactory cannot supply requested class
(Error=80040111)
Any ideas? I tried searching the net but only
found references to DivX.
Thanks
if you order from these guys the first thing to do
is delete the tftp entry. if you dont, your phone will disregard your
settings and continue to revert to sipphone settings.
default password is: admin
this is just for archival
purposes
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