On Tue, 16 Nov 2004, Peter Svensson wrote:
On Tue, 16 Nov 2004, Tobias Jönsson wrote:
On Mon, 15 Nov 2004, Jason Williams wrote:
After the Authenticte why not do a Playtones(Dial) this will give
dialtone
The dialtone won't stop after pressing first digit then. If course you can
have an X extension
hi john,
John Williams schrieb:
i made a patch that allows the compilation of chan_capi-0.3.5 against
current CVS-HEAD of asterisk.
If I remove the -2.95 from the CC declaration I get a very large number
of errors, the same ones I get when trying to compile without the patch.
ok, i forgot to
Off topic, but this seems like a good place to ask :)
Why do vendors think it is in their best interest to encrypt device config
files and then restrict access to the tools to create the config files?
snip
I'm sitting in a data center surrounded with probably close to $1,000,000
of big
Shaun Tierney wrote:
When I use the Dial command to connect a call using my Asterisk PBX, it
seems that the PBX says that the call was answered right when the two
channels are bridged together, rather than when the actually callee answers
their phone. I would like to be able to detect the actual
Who to generate ring tone to a calling party when the call is
passed
to an extension.
The asterisk answers correctly, plays welcome message and ring an
extension, but the caller does not here the rings.
Did you tell Asterisk to indicate ringing?
Asterisk will ALWAYS
Joseph wrote:
Who to generate ring tone to a calling party when the call is passed
to an extension.
The asterisk answers correctly, plays welcome message and ring an
extension, but the caller does not here the rings.
I'm willing to be you didn't set up Music On Hold correctly but are
using the m
List,
How about circumventing this dongle? Switches or PoE midspan units
that support forcing power on 4-5, 7-8 without detection? Found any
3rd party contraptions, like PoE splitters that tell the injector it's
ok, which can simply have an end crimped on in the right way to hit
the polycom or
Hey, thanks to everybody who posted to my earlier thread. Here's a
solution I came up with based on reading your scripts and advice.
It's really simple and stupid- but seems to work great. Incoming
calls for any type of extension can be configured to make winpopups
(or linpopups : )
On Thu, 18 Nov 2004 14:37:15 +1100, Adam Goryachev wrote:
Just wondering if anyone has used either of these motherboard with
a TE405p. My current board is causing problems, and I'm looking to
replace it...
gigabyte GA-7NF-RZ
gigabyte GA-7N400 Pro2
Thanks,
Adam
Not a direct answer I'm
Hi all,
I have a PBX working for a year with an Eicon Diva Server 4BRI. One
day it was a storm and nothing occurs, but after a a few days I can't send
and receive any calls. I have connected TEIs to Asterisk and other PBX and
when I try to dial, I hear correct tone two times, but then line
Hi
I have a particularly painstaking problem with ISDN
BRI. Currently Im running Fedora Core 2. I was running Red
Hat 9 (2.4.20-8 kernel) but the problem remains the same. My problem is
as follows:
I have a BRI ISDN card installed which is based on the HFC
chipset. I have had no
Hello,
I just purchased 10 G729 licenses for my asterisk box from Digium I was able
to register the key. But when i start asterisk it fails with the error
message:
[codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]: loader.c:248
ast_load_resource: Shared object libc.so.6 not found
The manual states that 'the wireless system supports automatic
registration' in the paragraph about registration of the users on the
callmanager.
[105]
id = 105
label = Z4040
callwaiting = 0
mailbox = 199
callerid= 105
If you're using CVS asterisk chan_sccp that's no
Hi all,
For some reason Music On Hold does not work. I have searched the internet for
solutions but found nothing that helped.
I use Asterisk 1.0.1 and mpg123 0.59r on Debian 2.6.7-1-386 (Sarge). mpg123
works on the commandline (I get sound from the soundcard). If I start Asterisk,
two
Hi everybody:
How much internet bandwidth and spees is enough to handle twenty simultanous SIP calls.
Do you Yahoo!?
The all-new My Yahoo! Get yours free!
___
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[EMAIL PROTECTED]
kido noagbodji [EMAIL PROTECTED] writes:
[codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]: loader.c:248
ast_load_resource: Shared object libc.so.6 not found
To run Asterisk under FreeBSD, using a precompiled module for Linux,
you're probably going to have to run a complete Linux
On Thu, Nov 18, 2004 at 03:27:40AM -0600, [EMAIL PROTECTED] wrote:
I just purchased 10 G729 licenses for my asterisk box from Digium I was able
to register the key. But when i start asterisk it fails with the error
message:
[codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]:
Actually, this is required to work for telco's (I would think this is
the same in most countries). Consider premium rate phone services (in
Australia, 1-900 xxx xxx) where you are charged $x per 'time unit'. eg,
$5/minute etc... The service operator is required to tell you how much
the
Hi ,
Anyone can help me to configure zapata.conf for a TDM04b ?? I can
place outgoing calls (with calls files) from Zap/1 but the problem is that
from Zap/ 3 is not possible. Is there a place to add more information
to configure channels ? Also, the channels configuration is static, I
mean if I
On Thu, 18 Nov 2004, Rich Adamson wrote:
Examples:
1. two-wire analog pstn lines: as soon as current draw is sensed by
the central office, answer supervision is generated by that central
office, period. It has nothing to do with whether * handled it or
whether an analog phone is hanging on
Hi
What type of expansion cards do you have on this board, i.e. how many fxo's
and how many fxs?
Regards,
Christiaan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ciprian Zetea
Sent: Thursday, November 18, 2004 12:02 PM
To: [EMAIL PROTECTED]
Subject:
I spent today trying to get openh323 working with Asterisk 1.0 on my new
AMD64 box. I ran into a number of problems. The first being that the
openh323 build scripts do not recognize x86_64 as an architecture that it
builds on. I hacked the scripts appropriately and got it built. I set the
Hello,
I'm still kinda new to asterisk, but I'm trying to setup the following
situation:
Aterisk running at port 5065, SER running at 5060 (done that, works fine)
Load of SIP clients registering at SER (no problem), SER routing the SIP
traffic to Asterisk (no problem).
Incomming:
*) Asterisk
Duane wrote:
Christopher Dobbs wrote:
What ever, Just trying to be a help to the system.
The benefit of enum over a relay service is the ability to interoperate
with others using SER and other VoIP PABXs as well, rather then being
limited to just other asterisk users, it's self managed via the
hi all.
ive got a problem implementing my own small office asterisk solution.
i want to use
- a hfc-card via mISDN in NT-mode to serve my siemens gigaset 3035 isdn
phone
- an avm b1 to connect to pstn
- sip, iax etc.
working:
- chan_capi via the b1 works fine, i can dial in and get the demo
-
Joost , I am running 2.6.8 SMP Sarge/Debian on a HP ml330 an have no
problem.
The Zaptel hardware is e T410P
Are you running without Zaptel Hardware ?
Jack
Joost Kraaijeveld wrote:
Hi all,
For some reason Music On Hold does not work. I have searched the internet for
solutions but found
Hi
I'm currently busy on a similar application with a hfc-card. However, my
needs is to interface the ISDN card in NE-mode with the operator. If tried
using Hisax but ran into a problem with the voice quality being bad in one
directions.
How did you manage to get chan_capi going for BRI ISDN
Hi Hammoud,
It all depends on the codec that you are using.
Best case scenario is with G723 codec 6.3Kbps per channel * 20, around 126K
without the overhead. But you problably won't be able to use this codec unless
you are in passthru mode (license is pretty expensive).
Using g729 you will
Hi,
I have one asterisk box with chan_oh323 to an external
carrier.
In order to have G.729 with this carrier, I use:
setGlobalVar(OH323_OUTCODEC=g729)
I have that asterisk box connected to another asterisk
box via IAX, with
disallow=all
allow=alaw
in the respective peer/friend chapters in
Hi All ,
Im planning on setting up Asterisk and was wondering if it was possible
to have it accept calls on a FXO card (TDM400b), then have it deliver
the call to a SIP based IP phone , or do I need to have a FXS module
and analog phones to accept calls via the FXO card?
Thanks ,
JK
As an additional info, G723 is like 1700 bytes per second, GSM is like
4600-4800 bytes per second as viewed by netstat - those numbers are with
ip overhead. I've been able to use a dialup as a link to get 2
simultaneous G723-based sip hardphones to * server.
kido noagbodji wrote:
Hi Hammoud,
Hi Jack,
[EMAIL PROTECTED] schreef:
Joost , I am running 2.6.8 SMP Sarge/Debian on a HP ml330 an have no
problem. The Zaptel hardware is e T410P
Are you running without Zaptel Hardware ?
Yep. I have two Winbond ISDN cards in the machine.
I am relieved that someone succeeded in running it
Hi
No Problem.
I've got a TDM400b board with 2 FXS boards and 2 FXO boards. I'm using the
fxo board in this fashion as you describe. However, I direct it to
voicemail. What you would do is the following.
In extensions.conf
exten = s,1,Dial(SIP/yourSipConfigName)
This should work fine.
Hi all,
I think there is no asterisk-addons version
for freebsd. Am I right? I tried to compile the standard version but I couldn't
do iton FreeBSD, may be the idea is as crazy as try to install asterisk
for linux on freebsd!
___
Asterisk-Users
Title: Zyxel Prestige 2002/2002L sound quality
Hi everyone,
I've been trying a Zyxel Prestige 2002L ATA with Asterisk, but I have a problem with very bad sound quality (using G711, it sounds very robotic and metallic) and there is a very long delay in the audio. This all doesn't happen with
hi christiaan.
Christiaan Brink schrieb:
I'm currently busy on a similar application with a hfc-card. However, my
needs is to interface the ISDN card in NE-mode with the operator. If tried
using Hisax but ran into a problem with the voice quality being bad in one
directions.
How did you manage
I'm using kern 2.6.5-1.358
I'll try out chan_misdn. Hopefully I'll have better luck with that.
Regards
Christiaan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Jagoditsch
Sent: Thursday, November 18, 2004 2:12 PM
To: Asterisk Users Mailing
On November 18, 2004 12:56 am, Duane wrote:
much nicer solution then the winpopup suggestion, I've actually been
looking for some tray bubble app like this for a while that was a mini
sip client, guess this will do in the mean time...
This is why I used the Jabber solution -- I already use Psi
Hi all,
I try to compile app_icd to test it but I can't compile it. I have
installed asterisk 1.0.2 and I download ICD and put files into
/usr/src/asterisk/apps/icd directory. I think that make.conf in icd
directory is ok but when I try to compile icd I obtain next error:
=== Compile:
The problem is that that should be dynamic :/
Take a look at this sip msg:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Max-Forwards: 10
Record-Route: sip:[EMAIL PROTECTED];ftag=as3f718642;lr=on
Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0
Via:
Not realy sure, but it seems that you are missing Zaptel timing,
Just Google for ZTDUMY or viop-org, ZTDUMY take TDM timming from a
USB-DEVICE
this might be your problem...
I am running in under root!
Jack
Joost Kraaijeveld wrote:
Hi Jack,
[EMAIL PROTECTED] schreef:
Joost , I am running
Joseph wrote:
When I call via * my bank and an automated system ask me to enter the
numbers they aren't recognized by the system. I'm getting an error
message that the numbers I entered are invalid. WHY?
1.) From what are you calling? A phone connected to an FXS port? A SIP
phone?
2.) How is
Title: Zyxel Prestige 2002/2002L sound quality
Hi,
This ATA works fine for us.
Haven't had any problem with sound issues, other than 1-way audio, but that
is another problem which is fixed.
But checked with different analog phones, and there seems to be a
difference. good and bad quality.
Hi all,
I was wondering if it is possible to connect 2 asterisk boxes together through
Zap channels.
I have 2 boxes with 2 TE410p cards per each box, what I would like to do is
connect the 2 boxes using Zap spans.
In one server I configure one span as output, and the other as input and I use
a
kido noagbodji wrote:
Hi Hammoud,
It all depends on the codec that you are using. Best case scenario is
with G723 codec 6.3Kbps per channel * 20, around 126K without the
overhead. But you problably won't be able to use this codec unless you
are in passthru mode (license is pretty expensive).
Hi all
If been working a while now trying to interface Asterisks
with BRI ISDN. Ive tried various drivers without any success. Im
running a HFC passive ISDN board in 2.6.5 kernel.
Is there anybody out there who have successfully interface voice
with Asterisk using BRI ISDN from their
Hello Christiaan
I have several * boxes running with AVM cards (Fritzcard, C2 and
Fritz-USB). Of course in TE mode interfacing with the operator.
Installation of card on SuSe is straightforward, other dists (including
debian, which we use) needs a little tweaking. * users chan_capi to
Hello Christiaan,
Thursday, November 18, 2004, 2:40:00 PM, you wrote:
CB Hi all
CB
CB If been working a while now trying to interface Asteriskswith
CB BRI ISDN. Ive tried various drivers without any success.
CB Imrunning a HFC passive ISDN board in 2.6.5 kernel.
are you using the
Title: Message
I have
3 HFC cards running in my PC in both NT and TE mode. Works well..
What problems are you having, what have you tried?
Rgds
Tim
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christiaan BrinkSent: 18 November 2004
i just do this:
tar -zxvf asterisk-1.0.0.tar.gz
but when i type make i get command not found
can some bathy help me hith this???
Rodney Acosta Coya.
Dpto. Tecnologia de la Informacion.
[EMAIL PROTECTED]
Tel:(53)(24) 62 611
-Mensaje original-
De: Jerry Geis [mailto:[EMAIL PROTECTED]
Over on the voip-info.org tiki I found this statement:
Mark (the man who made Asterisk PBX, www.asterisk.org) has an xbox that
has 4 analog ports via usb... aka the XBoxPBX
While I'm not interested in the xbox part of this, I wonder how one uses
USB for analog connections? Explanation? Pointer
Does anyone have an example of what the setup paramters
are
connecting asterisk with cisco call manager using a quad T1 card?
I see the other examples using SIP and CCM 4.0 but we dont have 4.0 yet.
I see other examples using h323 but we are not using that.
I dont see any configuration
I've tried using bri-stuff. I got everything compiled and installed but, it
seems that the driver stalls our ISDN NTU provided by the operator.
As I said I'm running the 2.6.5 kernel. I have a HFC card and our operator
uses alaw over EuroISDN. My zaptel.conf file is as follows:
loadzone=us
Does nobody else has got this Problem?
Or does nobody know how it should be fixed? ;-)
Bastian Schern schrieb:
Hi all,
I try to get the IAX2 peers from a MySQL DB. But this will not work. I'm
working with Asterisk 1.0.2.
First I had enabled MYSQL_FRIENDS in the channels Makefile, then I had
When you say command not found, is your system not finding make or is
make running and complaining that it can't find a Makefile ?
If the former, you may not have make (usually nmake) installed, or it is not
in your path. If the latter, you're probably in the wrong directory. Make
looks for
You can do a couple of things. If you are using a SIP device, or a device that
uses a PLAR code (Private Line Automatic Ringdown) then you can put that
number that you want to dial when the phone goes off-hook in that section.
That way, when you pick up the phone it will dial that number first
Title: Message
Hi Tim
Ive tried hisax. Got the
driver going with * but the problem is that the voice generated by Asterisk is
barely audible by the public operator user. Also, Hisax permits only the
use of one card.
I then resorted to zaphfc. Im
running the 2.6.5 kernel and with
i just do this:
tar -zxvf asterisk-1.0.0.tar.gz
but when i type make i get command not found
can some bathy help me hith this???
-Mensaje original-
De: Jerry Geis [mailto:[EMAIL PROTECTED]
Enviado el: Jueves, 11 de Noviembre de 2004 04:12 p.m.
Para: [EMAIL PROTECTED]
Asunto:
I had the same problem on Debian, the mpg123 in Debian is really mpg321 which
is supposed to be a drop in replacement. Well, I don't think it is, I
compiled mpg123-0.59r from source and it works now. You may want to give that
a try.
Pete
On Thursday 18 November 2004 04:32, Joost Kraaijeveld
i found de file Makefile but i dont now what to do with it
look at this
inux:/inst/pbx/asterisk-1.0.0 # make:make install
bash: make:make: command not found
linux:/inst/pbx/asterisk-1.0.0 # ls
. README.fpm apps callerid.c config.c editline
io.c pbx
Thats not even the newest firmware, 7.2 is newest. Isn't this illegal?
Matthew
- Original Message -
From: Kyle Hagan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, November 17, 2004 5:42 PM
Subject: [Asterisk-Users] Cisco
sounds like you dont have the make utility installed. is this an rpm based
system or debian or what? do this - which make
you should get something like
/usr/bin/make
if you get nothing, then you dont have make installed. if you dont have that
installed, there is probably a whole lot of
Hello,
I think you need to rewind a bit and do a bit of reading on how
applications are compiled on a *nix platform.
The Makefile is just a list of instructions for the make applications.
If you do not have the make application, and appropriate compilers and
libraries installed, you are going
Hello Tim
I'm struggling to get a HFC card running in NT mode. It seems to work
for a short period but then stops. The messages on the asterisk console
mentione something about event 6.
Is the zaphfc module enough to be loaded or must hisax also be loaded in
order to work?
Best regards,
On Thu, 2004-11-18 at 10:26 -0500, Rodney Acosta Coya wrote:
i found de file Makefile but i dont now what to do with it
look at this
[snip]
inux:/inst/pbx/asterisk-1.0.0 # make:make install
bash: make:make: command not found
linux:/inst/pbx/asterisk-1.0.0 # ls
.
[snip]
Try make make
Hi Pete,
[EMAIL PROTECTED] schreef:
I had the same problem on Debian, the mpg123 in Debian is
really mpg321 which is supposed to be a drop in replacement. Well, I don't
think it is, I
compiled mpg123-0.59r from source and it works now. You may
want to give that a try.
I use the real
my distro is novell linux desktop
yes is an rpm based system
look this
linux:/inst/pbx/asterisk-1.0.0 # - which make
bash: popd: directory stack empty
what can i do??
Rodney Acosta Coya.
-Mensaje original-
De: Chad Whitten [mailto:[EMAIL PROTECTED]
Enviado el: Jueves, 18 de
Kevin P. Fleming wrote
Noah Miller wrote:
IP 300's don't support PoE even though their brochures say
they do. Has
anybody have firsthand experience with them? Is this true?
None of the Polycom phones support PoE directly, but all of
them support
it via an external PoE adapter
I'm ordering some more phones - I have the Polycom IP 500's now and I
like them. I need some less expensive phones, and I'd like to stay
with all Polycoms for ease of administration. I've heard, though,
that
the IP 300's don't support PoE even though their brochures say they
do.
Has anybody
On Thu, 18 Nov 2004, Rodney Acosta Coya wrote:
my distro is novell linux desktop
yes is an rpm based system
look this
linux:/inst/pbx/asterisk-1.0.0 # - which make
bash: popd: directory stack empty
what can i do??
Is it possible that you are new to linux ? Try typing
which make
dont know if the novell linux desktop offers the devel tools with it. pop in
your install cd and see if you can find a make-version-number.rpm file. if
so, install it and whatever dependencies it asks for. if you cant find one
on the cd, my suggestion would be to find rpms of asterisk or get
Thomas Jagoditsch wrote:
hi christiaan.
Christiaan Brink schrieb:
[snip]
sorry, but i use chan_capi with the avm B1 card (this is an active
one, different then a A1/fritzcard), see my posting.
i see no way to use a hfc-based card or an avm A1/fritz with chan_capi.
so IMHO you need not chan_capi
Here is how I start asterisk with safe_asterisk:
[EMAIL PROTECTED] asterisk]# /usr/sbin/safe_asterisk 21 /dev/null
[1] 7514
[EMAIL PROTECTED] asterisk]#
[1]+ Done/usr/sbin/safe_asterisk 21 /dev/null
[EMAIL PROTECTED] asterisk]#
[EMAIL PROTECTED] asterisk]# !as
asterisk
Matthew Boehm wrote:
Thats not even the newest firmware, 7.2 is newest. Isn't this illegal?
Matthew
Definitely illegal, but 7.3 is the latest SIP firmware.
Jeb Campbell
[EMAIL PROTECTED]
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i found in my cd this files:
make-3.80-184.2.i586.rpm
makedev-2.6-403.2.i586.rpm
what do you think???
Rodney
-Mensaje original-
De: Chad Whitten [mailto:[EMAIL PROTECTED]
Enviado el: Jueves, 18 de Noviembre de 2004 09:59 a.m.
Para: Asterisk Users Mailing List - Non-Commercial
Ok Here you go.
Untar your file -xzf
Do a ./configure
Then make clean
Them make
Then make install
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I am getting a 6 second delay whenever i dial 9 to call someone using
PSTN, What could be causing this??
I am using the Wildcard X100p.
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To
Sean Kennedy wrote:
http://www.cdw.com/shop/products/default.aspx?EDC=568864
Can anyone tell me if this switch will be able to supply a Cisco 7940
phone with power? I've heard of PoE issues with differing switches and
the like, and I don't know how to check to see if this switch will be
able
Examples:
1. two-wire analog pstn lines: as soon as current draw is sensed by
the central office, answer supervision is generated by that central
office, period. It has nothing to do with whether * handled it or
whether an analog phone is hanging on the end at the customer's
location.
Kevin Blackham wrote:
How about circumventing this dongle? Switches or PoE midspan units
that support forcing power on 4-5, 7-8 without detection? Found any
3rd party contraptions, like PoE splitters that tell the injector it's
ok, which can simply have an end crimped on in the right way to hit
[EMAIL PROTECTED] wrote:
I am getting a 6 second delay whenever i dial 9 to call someone using
PSTN, What could be causing this??
I am using the Wildcard X100p.
Part of that delay is just waiting for the X100P to dial. Part of that
delay may be overlapping dialplan entries.
exten =
Hi,
Is there any method to log the reason a call was ended / terminated /
dropped??
I am getting a fairly high nimber of calls being dropped but have no way
of working out why.. I need to still upgrade Asterisk to ver 1.0 but I
still need a way to track the reason for the call dropping so that
David Gomillion wrote:
and ask... That was the only way I was able to get in touch with
PolyCom, and the way I was assured I would have the cables. If you're
interested, I can let you know when they come in if they were indeed in
the box.
Yes, that would be nice. In fact, once you know that is
[EMAIL PROTECTED] wrote:
I am getting a 6 second delay whenever i dial 9 to call someone using
PSTN, What could be causing this??
Pattern matching, perhaps?
What's your dialplan look like for the station you're calling from?
--
Andrew Thompson
http://aktzero.com/
List:
We are a Polycom reseller and all of the IP300's we have gotten have come
with the POE cable. If anyone needs further information please contact me
off list.
Garrett
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Gomillion
Sent: Thursday,
the commande is make;make install
not make:make install
you typed : instead of ;
Rodney Acosta Coya wrote:
i found de file Makefile but i dont now what to do with it
look at this
inux:/inst/pbx/asterisk-1.0.0 # make:make install
bash: make:make: command not found
Probably so. It's not like it's prohibitively expensive (AFAIK)
On Thu, 18 Nov 2004 08:33:16 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
Thats not even the newest firmware, 7.2 is newest. Isn't this illegal?
Matthew
- Original Message -
From: Kyle Hagan [EMAIL PROTECTED]
Perhaps your dialplan has another match possibility, and it's waiting for
the timeout to evaluate what you've dialed?
-Michael
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, November 18, 2004 9:15 AM
To: [EMAIL
Do a ./configure
Asterisk doesn't have a configure script.
-Seth
--
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559
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[EMAIL PROTECTED]
(clutches chest) *gasp* I dont have the newest???!? Ahh!! Oh look at that,
came out Nov 3rd. Geee..
Matthew
- Original Message -
From: Jeb Campbell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, November 18, 2004 9:08 AM
Hi
I'm having problems with Playtones. Playtones doesn't play
anything before I play something else.
When I enter this extension none of the Playtones are heard but
if I add a SayNumber(123) the last Playtones can be heard.
exten = s,1,Answer()
exten = s,2,Wait(2)
exten = s,3,Playtones(busy)
[EMAIL PROTECTED] wrote:
I am getting a 6 second delay whenever i dial 9 to call someone using
PSTN, What could be causing this??
I am using the Wildcard X100p.
___
probably some type of timeout, do you have more than one extension
beginning with 9
The Sipura 841 will support PoE. These are not shipping from Sipura until
the end of November, ready for early December. The Zultys Zip2 does not
support PoE.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent: Thursday, November 18, 2004
Kevin P. Fleming wrote:
Sean Kennedy wrote:
http://www.cdw.com/shop/products/default.aspx?EDC=568864
Can anyone tell me if this switch will be able to supply a Cisco 7940
phone with power? I've heard of PoE issues with differing switches
and the like, and I don't know how to check to see if
Messages like these dilute the
value of the Mailing list and draw attention away from valuable queries
So does your message, Brent.
Note that you don't answer the question in that it wasn't what ports
do protocols X,Y and Z use on asterisk? which would be easy to find
indeed. Jeff's post
[general]
static=yes
writeprotect=no
[default]
include = from-sip
include = outgoing
include = incoming
[from-sip]
exten = 1400,1,Dial(SIP/1400,15) ;phone1
exten = 1400,2,Voicemail(u1400)
exten = 1400,4,Hangup
exten = 1500,1,Dial(SIP/1500,15) ;phone2
exten = 1500,2,Voicemail(u1500)
exten =
not a good idea.. just get a tac account..
Jason
On Thu, 18 Nov 2004 09:08:57 -0600, Jeb Campbell [EMAIL PROTECTED] wrote:
Matthew Boehm wrote:
Thats not even the newest firmware, 7.2 is newest. Isn't this illegal?
Matthew
Definitely illegal, but 7.3 is the latest SIP firmware.
Jeb
/SNIP/
Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps
on the IP level without silence suppression because of the additional
overhead imposed by protocols like RTP, IP, etc . If you add the
Ethernet (or WAN protocol overhead) this will increase even more
(although
http://www.grandstream.com/b21p1.0.5.18.zip
I can't get it to call out, but many people have been successful with it.
___
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all my 7960's work great. loading the new sip image was no problem ,
took about 5 min. I even got them from ebay =
Jason
On Wed, 17 Nov 2004 14:52:32 -0800, Tracy R Reed
[EMAIL PROTECTED] wrote:
On Wed, Nov 17, 2004 at 05:07:33PM -0500, Bob Willock spake thusly:
I just bought a couple of
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