Hi all,
Is there a way to put a voicemail quota to a SIP user? I mean a quota on the
user's mailbox instead
of a particular message of the user like the 'maxmessage' does currently.
Quata can be total file size of message or
total minutes of messages of a mailbox.
Any help or suggestions?
Just to follow up on my post of yesterday, the solution was simple (thanks
to the asteriskTFOT book!)
Simply add the following line (modified, of course!) to the call file:
CallerID: Asterisk 800-555-1212
Regards,
Keith
- Original Message -
I'm calling people on Zap interface using
Hello,
did you try using a Local/XXX channel? it should work!
l.
On Tue, 01 Nov 2005 15:10:03 +0100, Stefan Günther
[EMAIL PROTECTED] wrote:
Hi,
I want to transfer a call that has come into one queue, and that I have
already accepted, into another queue.
When I try this asterisk tells
Title: Installing beta2
Once built no matter whether I do make install or make clean I get the same output
[EMAIL PROTECTED] asterisk]# make clean
build_tools/make_version_h include/asterisk/version.h.tmp
if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo;
What chipset that card use??
Pedro Nunes
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj
Sent: terça-feira, 1 de Novembro de 2005 23:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Fritz!Card PCI
There are lots of different ways to accomplish the same thing in *,
so there is no way to answer your should do question without
looking at what you've defined. You're obviously doing something
wrong if you can't get any provider to work, and no one is going
to be able help identify what you're
hi all;
Any one have experiment with this Ericsson MD evolution and asterisk,
i try to do that:
Phone-PABX Ericsson MD evolutionBox with asterisk server and TE110P
When i try to make call with my phone behind the Ericsson PABX, i had
just 4 digit in my asterisk!!!
Thanks
Is it possible to connect to Asterisk from an external application?
What I mean, to connect and execute its own extensions, created by
some other program:
exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller)
or
exten = $NUMBER_I_WANT,1,txfax($FILE_I_WANT|caller)
and Asterisk will dial
On Wed, 2005-11-02 at 11:29 +0100, Tomasz Chmielewski wrote:
Is it possible to connect to Asterisk from an external application?
What I mean, to connect and execute its own extensions, created by
some other program:
exten = 1234567,1,txfax(/home/steveu/testfax.tif|caller)
or
exten =
Hello!
First problem with 1.2-beta2.
All I hear during Echo() is noise. No matter which codec selected.
However, when using ulaw noise sounds better than g723 :)
My equipment is Sipura SPA-3000. Works fine with 1.0.9 amd 1.0.7.
___
--Bandwidth and
Thanks for this one, Greg !
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: mardi, 1. novembre 2005 16:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MTP required for CCM integration ?
You
Hi Dan,
Your comments definitely help. Thanks a lot.
I'll probably have more remarks / questions early next week.
BR, - Patrick -
From: Dan Austin [mailto:[EMAIL PROTECTED] On
Behalf Of Dan Austin
Sent: mardi, 1. novembre 2005 20:18
To: Asterisk Users Mailing
Hi,
anyone has a working example of this new function ?
that's all that I have found
-= Info about function 'REGEX' =-
[Syntax]
REGEX(regular expression data)
[Synopsis]
Regular Expression: Returns 1 if data matches regular expression.
[Description]
Not available
Tnx!
--
Best regards,
Title: Options for 3-way or Conference Calling
Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling or conference calling. My goal is to keep this as simple for my users as it was with our legacy PBX. On our old phone system, a user
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it.
On 11/2/05, Dave Morrow [EMAIL PROTECTED] wrote:
Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling
Are you installing over a previous source tree? If so, please rm -rf the previous source tree and install the new source tree from scratch.
On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote:
Once built no matter whether I do make install or make clean I get the same output
[EMAIL PROTECTED]
On Wed, 2005-11-02 at 07:16 -0500, BJ Weschke wrote:
Yes. I believe the Cisco phones do conferencing in the same fashion.
I'm not 100% on whether or not the SPA-841 or the new SPA-941 does it.
If not there is always features.conf :)
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK
Hi, I had removed all old versions before starting and
downloaded from CVS.
Regards
Lee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BJ
WeschkeSent: 02 November 2005 12:20To: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users]
Hello all,
We'd like to use asteriek as an internal pbx connected to an external sip
provider to make outbound/inbound calls to pstn.
We have the provider and have installed an asterisk at the office.
Does anyone have a sample config?
We need 25 telephone numbers(dids), to be registerd to the
Can you please suggest me some graphical interface (like AMP)? I have tried to
install AMP but I have some problems and on AMP forum and mailing list I didn't
get answer.
Two things I need to have are.
- list of calls for every user.
- some information about Linux (processor load, HDD, network
On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote:
Hello all,
We'd like to use asteriek as an internal pbx connected to an external sip
provider to make outbound/inbound calls to pstn.
We have the provider and have installed an asterisk at the office.
Does anyone have a sample config?
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case
It seems to be what I needed
Thanks for help.
Best regards,
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de trixter aka
Bret McDanel
Envoyé : mercredi 2 novembre 2005 14:09
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re:
Hunt, Bill wrote:
While I don't disagree in principle, I think an issue is that
much of the benefit of this list is the knowledge gained by
reading about other people's problems and resolutions. If these
discussions start being held in other languages we will not all
be able to benefit
[EMAIL PROTECTED]
- Original Message -
From: Tomislav Parčina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 02, 2005 7:41 AM
Subject: [Asterisk-Users] Graphical interface
Can you please suggest
Previously I would get two events on my Zap channel which indicated
ringing and answered. Now I am getting polarity reversal events:
Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17
(Polarity Reversal)...
Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got
Can you issue the following command on FC3 and let us know the results?
rpm -q kernel-source zlib zlib-devel openssl openssl-devel
On 11/2/05, Lee Archer [EMAIL PROTECTED] wrote:
Hi, I had removed all old versions before starting and downloaded from CVS.
Regards
Lee
Thankyou, this was a great primer for me also.
Chris
trixter aka Bret McDanel wrote:
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the
Thank you, this is definitely an option. Right now I'm trying to make something
work on my Linux installation (FC4). And I like to install as much things on my
own, so that I really can se how that stuff works.
Tomislav
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
please remove me from your mailing list. [EMAIL PROTECTED]asterisk [EMAIL PROTECTED] wrote:
http://lists.digium.com/mailman/create
This list supports English (USA). Possibly our spanish speaking friends need their own list?
Thanks,
Steve
- Original Message -
From: Carlos
Hi it says
[EMAIL PROTECTED] ~]# rpm -q kernel-source zlib zlib-devel openssl
openssl-devel
package kernel-source is not installed
zlib-1.2.1.2-3.fc3
zlib-devel-1.2.1.2-3.fc3
openssl-0.9.7a-42.1
openssl-devel-0.9.7a-42.1
Which is odd cos the sources are installed. I'm using the 2.6.9-1.667
On Wed, 2 Nov 2005, [iso-8859-2] Tomislav Parčina wrote:
Thank you, this is definitely an option. Right now I'm trying to make
something work on my Linux installation (FC4). And I like to install as much
things on my own, so that I really can se how that stuff works.
Then I guess you'll
On Wed, Nov 02, 2005 at 01:41:38PM +0100, Tomislav Parčina wrote:
Can you please suggest me some graphical interface (like AMP)? I have tried
to install AMP but I have some problems and on AMP forum and mailing list I
didn't get answer.
Two things I need to have are.
- list of calls for
Title: extension
I would like to know how to set up will be
in one sipura 2002 box and have another same
Extension but in different locations like bedroom and kitchen I believe I need
two sipura boxes are need it. Can you help?
I tried it but nothing happens, seems asterisk is not getting the dtmf or
something and I get an error on the CLI saying something like this when the
keys are pressed.
-- Attempting native bridge of SIP/212-227a and SIP/201-dc6b
-- Native bridge of SIP/212-227a and SIP/201-dc6b was
Previously I would get two events on my Zap channel which indicated
ringing and answered. Now I am getting polarity reversal events:
Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17
(Polarity Reversal)...
Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got
If the two phones are going to have the same extension its just a matter
of wiring two phones to one port on the sipura unit.
If you disconnect the telco service from your house's internal phone
wiring and plug one of the Sipura ports into a phone jack, that sipura
device will provide power
How can I configure Asterisk to tell me if there
are messages on my voice mail as soon as I hook up an internal
phone?
Regards,
Andrea
Frigo
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
I tried to open the spanish list, but the mailmain server didn't let me
open, because didn't recognize my password.
So, until we can do that JODERSE.
He tratado de abrir la lista en español, pero el administrador de la lista
no me lo ha permito, ya que no reconocio mi contraseña.
Por lo tanto,
This works for me, your mileage may vary:
in sip.conf add these two lines under the sip user:
mailbox=/[EMAIL PROTECTED]
/notifymimetype=application/simple-message-summary
example for mailbox=
[EMAIL PROTECTED]
The SIP device must support this feature of course.
And if you're not using SIP
Does anyone know if the intel e7230 chipset in the new dell poweredge
sc430 and poweredge 830 servers is compatible with the te110p and
tdm400p cards?
I know there were problems with previous generation dells, but I've
read that these work fine. Can anyone confirm this?
thanks
robbie
Mark Edwards wrote:
This indicates that 602 is a dynamic host. It must therefore register
with the pbx so that the pbx knows where to send data.
In this state it is unregistered so it will be unlikely you can call it.
That was I expected, that I cannot call it, but I could
That gives
Ever since I started playing with Beta versions of Asterisk, I've had a
problem. It might just be coincidence though, since before that I didn't
touch the Asterisk PC for a good 2 weeks and I had done alot playing around
with config files.
I have a 4 port FXS/FXO card (with 2 of each
For analog phones - same thing 8-) except it is in zapata.conf
mailbox=whatever ; under (er just above) the channel.
Should give you a stutter dial tone.
Brett
On 11/2/2005, Adam Moffett [EMAIL PROTECTED] wrote:
This works for me, your mileage may vary:
in sip.conf add these two lines under
I am trying to figure out how to setup asterisk with a TDM400 (TDM04B), so
that the first 3 lines incoming
will be answered and the 4th line is just for outgoing
calls but doesnt answer on incoming calls.
The easiest way to do that is to give the channel a weird context name.
For
You should receive a short ring every 5 or
10 minutes if you have voicemails on your box.
Now, if you have an IP Phone, you can have
a led (Like the Cisco 7960, or an icon like on the Swissvoice IPS-10) that
reports you that condition.
Regards,
Carlos Alperin
From:
Just started getting this warning message about every minute.
ast_sched_runq ran 20 scheduled tasks all at once
I know it's a warning but Mark/Kevin Co must have thought it worth
mentioning.
So its a patch from me, that I may regret.
I'd strongly suggest leaving it in there for
Previously I would get two events on my Zap channel which indicated
ringing and answered. Now I am getting polarity reversal events:
Nov 2 07:01:25 NOTICE[18246]: chan_zap.c:6014 ss_thread: Got event 17
(Polarity Reversal)...
Nov 2 07:01:28 NOTICE[18246]: chan_zap.c:6014 ss_thread:
Hi,
I have installed the asterisk 1.2 beta version and I
have created the voicemail table described on this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail,
but when I start the asterisk server I receive the
following error.
Any idea ?
Thank you
Looking for a Current Dell model, tower or 2U rackmount, that
has (3)
usable PCI slots? Was just cruising Dell.com and can't find a
detailed
spec on any of the server offerings that tells me the number of
PCI slots
available. Anyone using Dell for PBX builds can point me in the
right
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax
number to the fxs port to which the fax machine will be connected.
I believe this will work, but wanted to know if anyone has done this.
Do I need to set faxdetect=both in zapata.conf?
I am assuming that Asterisk will
On Wednesday 02 November 2005 11:17, Kevin Hanson wrote:
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax
number to the fxs port to which the fax machine will be connected.
Do I need to set faxdetect=both in zapata.conf?
I am assuming that Asterisk will bridge between
Hi,
I am planning to connect my Asterisk PBX to one or two POTS lines, and
am wondering if it is better to use a TDM card for this, or one or two
SIP devices with FXO ports on them (such as an SPA-3000, Grandstream
488). I am interested in voice quality and reliability of operation and
am
Anton Krall wrote:
I tried it but nothing happens, seems asterisk is not getting the dtmf or
something and I get an error on the CLI saying something like this when the
keys are pressed.
-- Attempting native bridge of SIP/212-227a and SIP/201-dc6b
-- Native bridge of SIP/212-227a and
For everyone that had inquired about the Find-Me/Follow-Me
application, it's now up in the bug tracker at
http://bugs.digium.com/view.php?id=5574.
It should compile cleanly against a 1.2b2 install.
On 10/14/05, BJ Weschke [EMAIL PROTECTED] wrote:
CF -
You're right. Most of this can be
We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI slots.
On 11/2/05, Tom Rymes [EMAIL PROTECTED] wrote:
Looking for a Current Dell model, tower or 2U rackmount, that
has (3)
usable PCI slots? Was just cruising Dell.com and can't find a
detailed
spec on any of
We have built an Asterisk network using an MPLS-based IP VPN. We have
one location in New Brunswick Canada that consistently gives us major
quality problems, whereas the others are flawless. Quality problems
take the form of static, poor voice tonality, popping clicking, drops,
sporadic echo,
On Wednesday 02 November 2005 11:42, Kevin P. Fleming wrote:
-- Attempting native bridge of SIP/212-227a and SIP/201-dc6b
-- Native bridge of SIP/212-227a and SIP/201-dc6b was unsuccessful
-- Attempting native bridge of SIP/212-227a and SIP/201-dc6b
-- Native bridge of
We have a few satellite trunks for VoIP in Africa and have some experience.
Please mail me off list and we can discuss it
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: den 2 november 2005 18:01
To: Asterisk Users
Adam,
I personally think that replacing hard-wired network and going with Sats is
a mistake. Judging from pure round-trip delay you measured the packet round
trip seems sufficient to have a good conversation, but pinging is not enough
to trouble shoot the network problems. You will need to do a
I would like to manipulate phone call direction to voicemail for lunch,
after hours etc, but am unsure how to do this. Could someone
point me to a howto or quickly explain the concept?
Thanks
Neri
___
--Bandwidth and Colocation sponsored by
I have no experience in the matter whatsoever ;)
But, I can say that long distance phone calls (non-voip) are sometimes
carried over sattelite when fiber is not available.
It must be possible for voip, but the latency and jitter would be
tremendous and although I am not an expert on the
Are you having the same problems under terrestreal links ? which codec
do you use, are you using a dedicated channel on the vsat for it to
take the upstream load ?
Whats your jitter settings ?
On 11/2/05, Adam Moffett [EMAIL PROTECTED] wrote:
I have no experience in the matter whatsoever ;)
I just went through the same thing.
I settled on the GoToIfTime application. One strange thing about
GoToIfTime is that it doesn't allow an else argument, so you'll need a
sequence of if's to get things done.
try something along these lines:
[yourcontext]
;lunchtime
exten =
Robbie Hughes wrote:
Does anyone know if the intel e7230 chipset in the new dell poweredge
sc430 and poweredge 830 servers is compatible with the te110p and
tdm400p cards?
I know there were problems with previous generation dells, but I've
read that these work fine. Can anyone confirm
AudioCodes is widely available.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan
Sent: Tuesday, November 01, 2005 7:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] server hardware
On Tue, Nov 01,
BTW, show application GoToIfTime in the CLI will tell you the whole
syntax. It can also take days of the week and I think months of the
year as arguments, but that wasn't an issue for me since we're 7 days a
week.
Adam Moffett wrote:
I just went through the same thing.
I settled on the
Matt wrote:
We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI slots.
On 11/2/05, Tom Rymes [EMAIL PROTECTED] wrote:
Looking for a Current Dell model, tower or 2U rackmount, that
has (3)
usable PCI slots? Was just cruising Dell.com and can't find a
detailed
spec on
On Wed, 2005-11-02 at 16:56 +0100, Luca Lafranchi Lists wrote:
Hi,
I have installed the asterisk 1.2 beta version and I have created the voicemail table described on this page http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail,
but when I start the asterisk
I would like to manipulate phone call direction to voicemail for lunch,
after hours etc, but am unsure how to do this. Could someone point me to a
howto or quickly explain the concept?
I would recommend checking a database value over the time based
GoToIfTime unless you are always go to and
Hi,
sorry - I know that problem is not directly related to asterisk but mabe
someone can help anyway.
After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is
mostly not possible to dial numbers with leading zeros like 0018...
If you do so you see on the diplay an number
Andrew Kohlsmith wrote:
On Wednesday 02 November 2005 11:17, Kevin Hanson wrote:
I have a TE110P and a TDM10B. Via DID, I want to route calls to the fax
number to the fxs port to which the fax machine will be connected.
Do I need to set faxdetect=both in zapata.conf?
That would be great. Thank you.
Robbie Hughes wrote:
Does anyone know if the intel e7230 chipset in the new dell poweredge
sc430 and poweredge 830 servers is compatible with the te110p and
tdm400p cards?
I know there were problems with previous generation dells, but I've
read that these
Rene Nelson wrote:
I would like to manipulate phone call direction to voicemail for
lunch, after hours etc, but am unsure how to do this. Could someone
point me to a howto or quickly explain the concept?
include = atlunchcontext|11:00-11:59|mon-fri|*
include =
Did you ever find a solution for this problem? I have it on latest Beta 2
Bart
- Original Message -
From: Walt Reed [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, October 21, 2005 7:26 AM
Subject: [Asterisk-Users] Double DTMF with tdm card
I have a TDM22B
We have a poweredge 2850 that we use for our VoIP server and it has 3 PCI
slots.
I'm greeting to hear this. I have installed some Digium cards into this
kind of servers.
I get surprised when the slots pci gets shared IRQ with ethernet
devices, raid controller or VGA card.
Anybody knows
Bump - I'm stuck until I can find a solutions
Please help - I'll try anything!
Bart
- Original Message -
From: Bart Fisher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November 01, 2005 5:37 PM
Subject:
On Wed, Nov 02, 2005 at 12:31:48PM -0500, Adam Moffett wrote:
I have no experience in the matter whatsoever ;)
But, I can say that long distance phone calls (non-voip) are sometimes
carried over sattelite when fiber is not available.
It must be possible for voip, but the latency and jitter
Morel Mosolff wrote:
Hi,
sorry - I know that problem is not directly related to asterisk but mabe
someone can help anyway.
After updating our polycom ip 500 sip phones from 2.6.1. to 2.6.2.0032 it is
mostly not possible to dial numbers with leading zeros like 0018...
If you do so you see
Andrew Kohlsmith wrote:
I've been thinking of a way to get across the idea that a native bridge was
unsuccessful in more friendly terms for a bit but nothing really concise
has come to mind. Some reason code might be handy... unable due to
differing codecs, unable due to necessity to listen
I'm having some issues, and thought it wise to check with the list before
putting in any more time
Here we go:
1) Do Zaptel BRI (Cologne based cards) support DID routing (I believe they
do, but the behavior of my (*) server is making me doubt, and I want to be
sure before attempting any more
On Wednesday 02 November 2005 13:09, Kevin Hanson wrote:
Did you have to set 'echocancel=no' or fiddle w/ any other echo related
settings in zapata.conf for that channel?
No; the echo canceller is automatically disabled upon reception of a 2100Hz
tone (which is part of the start of all modem
Hi all,
I'm looking for a fax solution with Asterisk. I would like the users to be able to hook up regular fax machines to their SIP ATA's and send/receive fax from PSTN and/or other SIP clients.
My goal is:
fax machines - SIP ATA - Asterisk - T1(TE406E) - fax on PSTN
It looks
include = atlunchcontext|11:00-11:59|mon-fri|*
include = notatlunchcontext|09:00-10:59|mon-fri|*
include = notatlunchcontext|12:00-18:00|mon-fri|*
include = afterhourscontext|18:01--8:59|mon-fri|*
I wasn't aware that include allowed a time qualifier. Does that mean
that the specified
On Wednesday 02 November 2005 13:28, Kevin P. Fleming wrote:
For now I have removed the message (I added it recently), since it isn't
accomplishing what it was supposed to.
No problem, but it would be very handy to see the bridge status through show
channels type of output. a Bridge Type
Sattellite links aren't cheap, and, the worst of all, you have in a idel
condition, 1.4 seconds latency.
Hope this help...
Juan.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Adam Robins
Enviado el: Miércoles, 02 de Noviembre de 2005 02:01 p.m.
Para:
According to http://www.extremetech.com/article2/0,1697,1880749,00.asp
ATI is delivering a GPU enabled transcoding method that cuts video
transcoding down to 1/5 the time it would take the cpu. This might also
be applied to audio codecs in theory (I havent looked into it enough).
Lets face it
I am in the US, NYC using a TDM400 card. I never have never seen this
issue until now. I see some code has been changed in this area recently.
MARK.
Rich Adamson wrote:
Previously I would get two events on my Zap channel which indicated
ringing and answered. Now I am getting polarity
Title: OS for ABE
We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only supports Fedora 3 and Red Hat EL3, we typically use CentOS. Our problem with this scenario is that RHEL3 is an
On Wed, 2005-11-02 at 16:56 +0100, Luca Lafranchi Lists wrote:
Hi,
I have installed the asterisk 1.2 beta version and I have created the
voicemail table described on this page
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail,
but when I start the asterisk server I receive the
On Wednesday 02 November 2005 14:11, trixter aka Bret McDanel wrote:
According to http://www.extremetech.com/article2/0,1697,1880749,00.asp
ATI is delivering a GPU enabled transcoding method that cuts video
transcoding down to 1/5 the time it would take the cpu. This might also
be applied to
On Wed, 2 Nov 2005, Juan Janczuk wrote:
Sattellite links aren't cheap, and, the worst of all, you have in a idel
condition, 1.4 seconds latency.
I know you can get less, our client in the mid-west uses Hughes with under
600ms. But never attempted to do VOIP over it.
--
I'm running Asterisk 1.2.0b2 (also tried latest CVS HEAD) in my lab and
i've come across a strange problem.
I've setup an extension to call the meetme application, when i call that
extension it functions as expected, informing me of my conference number
and that i'm the only one in the conference
Weuse Fedora 3 and ABE-A.1
The pair has been workinggreat for usso far.
AK
On 11/2/05, Eric Alexander [EMAIL PROTECTED] wrote:
We are setting up ABE for a client of ours. This is not our first Asterisk install, far from it, but it is our first time using ABE. Here is the problem, ABE only
Rene Nelson wrote:
I would like to manipulate phone call direction to voicemail for lunch,
after hours etc, but am unsure how to do this. Could someone point me
to a howto or quickly explain the concept?
Thanks
Neri
Hi Neri,
The command GotoIfTime() if your answer here.
See
Price is high that is correct but latency is not correct. We have a number
of Satellite VoIP Trunks in Africa and no location has more then 500 ms
latency. In all locations we have 2 Mbit dedicated lines using C-band and
the hub is in the US. But price is HIGH. 6000 usd per month
Anders
Your ABE purchase comes with Digium support for Installation. You
should call them for the answers to your questions.
On 11/2/05, Eric Alexander [EMAIL PROTECTED] wrote:
We are setting up ABE for a client of ours. This is not our first Asterisk
install, far from it, but it is our first time
-Original Message-
From: Carlos Chavez [mailto:[EMAIL PROTECTED]
Sent: mercoledì, 2. novembre 2005 19:04
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voicemail in Realtime mode
On Wed, 2005-11-02 at 16:56 +0100, Luca
Stephen Arulraj wrote:
Anyone knows how I can use this ISDN card for asterisk as a BRI trunk
interface?
Thanks,
Stephen
Hi Stephen,
Is this a new version of the AVM card? If not (or even if it is), you
may find the following pages helpful:
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