Im publishing tftp through my firewall to support
external Cisco 7960 sip phones. I know that the primary port is 69 for tftp.
However, tftp also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad
range) In an effort to limit the secondary ports that are opened, some Windows
If you have an account you can try: http://control.sixtel.net This works
and they seem to be adding some features. My service still works.
However sixtel has been unable to tell me how much $ is available for
use. I'm not too confident at this point.
-Original Message-
From: [EMAIL
In the example below if I dial valid extension 1000, the Invalid
context plays pbx-invalid as it is included with _7 context.
Include voicemail in the main context.
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Asterisk-Users@lists.digium.com
I have a need to have the two sip phones register with the same
extension (at least I think I have the need :)
Consulting the wiki about the dialplan and the dial application
reveals that you can dial several phones at once, or in series,
whichever you wish.
Dial(SIP/2000SIP/2001) will do the
One way to do this would be to create a call queue with the two sip phones
as separate extensions connected to the one logical extension (the queue).
The other, and possibly simpler way to do it is to use
Dial(SIP/extensionSIP/extension) to ring both sip phones at the same
time. Regardless, you
Please see comments inline.
Rusty Shackleford wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald Wiplinger
Sent: Tuesday, July 26, 2005 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ASTCC:
On Wed, Aug 03, 2005 at 12:38:17AM +0200, Michel Koenen wrote:
I have this working with a Teles ISA card, see config below (numbers
are changed because I dont want everybody to call me;-) )
In modem.conf
ZapHFC is configured in zapata.conf, not in modem.conf, right?
--
Tzafrir Cohen
I've got two pa1688 phones that I want to set up to communicate between
branch offices without a gatekeeper. Both phones will be behind a
firewall and I want to use port forwarding so the phones can communicate.
Are you using these phones with SIP? Why not try IAX2?
I tested the phones
I have IAX2 (FWD) partially working. I can place calls from my
Asterisk box but I cam unable to receive them (comes back as
busy). I have my firewall forwarding the udp ports 5060, 4569,
5036 and 1 thru 2 to my asterisk server. I think I have
the firewall correctly setup as I can
On Tue, Aug 02, 2005 at 05:47:50PM -0400, Tim King wrote:
# It must be in the module loading order
# Span 1: WCTDM/1 Wildcard TDM400P REV I Board 2
fxoks=1
fxoks=2
fxoks=3
fxoks=4
# Span 2: WCTDM/2 Wildcard TDM400P REV I Board 3
fxsks=5
fxsks=6
fxoks=7
the extension register ok on asterisk server , but not audio is transmited
on answer a call
look for canreinvite=no in sip.conf
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To
Hello. I have a linux and two sip-ata's, a sipura 2002 and a GS ht-386. I
also have three sipphone numbers. I can connect the atas to the sipphone
accounts and I get a dial tone and I can call my house and it says, Thank
you for using SipPhone...
Using asterisk, I have the ata's
can you give me more details ? like :
are you using one asterisk server in public ip and two phones behind NAT or
two asterisk servers both are behind NAT and haveing phones connected
locally one with each other...
after that i can help u
- Original Message -
From: Oliver Bode [EMAIL
On Tue, Aug 02, 2005 at 10:46:17PM -0700, Chad Brown wrote:
I'm publishing tftp through my firewall to support external Cisco 7960
sip phones. I know that the primary port is 69 for tftp. However, tftp
also uses secondary ports ranging from 1,0XX to 30,XXX. ( A broad range)
In an effort to
hi
but i don't think IAX2 is good, because with IAX2 RTP packets goes via IAX
servers as mini packets not directly from one client to other client so for
a big implementation it may consume more bandwith then that of a SIP
solution
rest is up to the user...
- Original Message -
From:
hey chad,
just a heads up tftp is one of the worst protocols to use
when your behind a nat or firewall it drove me pretty crazy a while
ago.
Carlos AlcantarRace Technologies, Inc.101 Haskins
WaySouth San Francisco, CA 94080P: 650.246.8900F: 650.246.8901E:
carlos at race.com
From:
Hi,
We are trying to set up
an asterisk configuration using some 7960 Cisco Telephone. We need to deploy
those in our company and we also need to see on the screen who is on line or
not. After making a research on the web, we thing that we have to use MGCP or
sccp.
Does anybody have
I'm similarly exacerbated over the FXO PCI Master Abort thing. Right
now, I'm totally stuck!
I dont have much more info to give, but I'm SURE somebody on this list
is running a X101P card (ambient md3200), on linux. I can't see how
they can have failed to come across the same problem - since
Hello Tim,
I am definitely interested in testing
it.
Please contact me off the list.
Best Regards,
Boris.
If anyone is interested I'm (slowly) developing a
GPL'd Java applet that works as an IAX softphone.
I should have a test version out at the end of the
week for a limited number
I understand. However, Im
successfully managing this without any problems using a Windows tftp server by www.winagents.com. This software allows
you to limit secondary transfer connections to a range of IPs. Therefore you
only need to open up port 69 and the range you specify. Everything
Hi everybody,
I managed to install card over Suse 9.2, I
substituted Zaptel drivers and compiled them. Now "ztcfg" says I have one card
with correctly configured 31 channels, but red led on back of card doesn't
flash. Suse 9.2 has detected the card as a Tiger Jet card, since the chip on it
Hi Matthew,
i found the following link very usefull:
http://www.orderlyq.com/asteriskqueues.html#moh
It is an alternativ to mpg123. It works very fine for me.
Regards
Matthew Boehm wrote:
OK. So I did a test last night. All of asterisk's threads where using
0.0% CPU.
I made 1 call to
U can use this way in extensions.conf:
exten = 2,1,Dial(${BRUNO_FGA}${GIORGIO_FGA},${RING_TIME}) ; supp-tecnico
Bruno
Kevin Hanson wrote:
I have a need to have the two sip phones register with the same
extension (at least I think I have the need :)
A client wants an incoming call to ring
u can use this:
exten = i,1,Playback(invalid_selection)
exten = i,2,Goto(inbound_menu,_X.,1)
Bruno.
Joseph wrote:
Ho do you folks solve the problem with invalid extension when someone
dials a wrong number?
For example if somebody dial prefix _7 I want to allow tall free
numbers from that
Ashish Raikwar wrote:
can you give me more details ? like :
are you using one asterisk server in public ip and two phones behind NAT or
two asterisk servers both are behind NAT and haveing phones connected
locally one with each other...
after that i can help u
- Original Message -
Try to control the file in the server... i have seen that this phone
change the server file in an wrong way...
Bruno.
Brent Davidson wrote:
I have a Polycom Soundpoint IP 500 that I have been using with Asterisk
for a few weeks. It has been working OK, no major problems other than a
freeze
Gmail users,
I had the similar problem, but I discovered that all my mail for 30-31 july
was delivered into my junk folder. Then I selected them all and move then to
the inbox. Since then I have been receiving mail from the list
goksie
-Original Message-
From: [EMAIL PROTECTED]
Matt Brooks wrote:
I am just emailing to inform you guys that a new website has been
created for asterisk.org. You can find the beta site up at
http://beta.asterisk.org. It utilizes the drupal portal framework and
Looking very good and much easier to navigate! Great work!
Cheers,
Kristof
Kevin, can I make a suggestion that you look at ring groups (possibly
even download [EMAIL PROTECTED] - as you can implement ring groups really
easy using AAH).
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruno De
Hi,
Has anyone come up with a clever way of indicating DND is activated?
I've thought of stutter dial tone and using the mwi, but have no idea
how to implement these. I'm using Budgetones. My concern is that users
will activate the DND, then forget about it not realizing that they are
not
Chad Brown wrote:
I'm publishing tftp through my firewall to support external Cisco
7960 sip phones.
I hope the files requested by the Cisco phones don't contain username
/ password information. Passing that in cleartext is just so wrong ;-)
--
Andreas Sikkema bbned NV
Hi,
Can anyone give me any information at all to get app_intercept working?
I've found these pages, but there is just not enough for me to get it going.
http://www.pbxfreeware.org/archives/2005/06/new_download_--.html
and
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692
I have this working with a Teles ISA card, see config below (numbers
are changed because I dont want everybody to call me;-) )
In modem.conf
ZapHFC is configured in zapata.conf, not in modem.conf, right?
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
Yes, I know but I gave the
Darren's suggestion did the trick, thanks.
Keep up the good work!!!
Ade.Darren Wiebe [EMAIL PROTECTED] wrote:
You should have your pattern set to ^4207. Then the pattern has to start with 4207. The way my setup would be is ^0114207.Darren Wiebe[EMAIL PROTECTED]Ade Agbero wrote: Astcc applies a
Mark Phillips schrieb:
I downloaded the latest CVS a few days ago. It all compiled nicely on my
new AAH platform. However, it won't start up.
Investigation of my log files produces this;
Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so]
Jul 26 22:59:18 VERBOSE[31473] logger.c:
Nkm [EMAIL PROTECTED] :
On 8/2/05, Darren Wright
wrote:
Can anyone point me to the location of the 7970 SIP image? I'm logged
There's no SIP firmware for 7970, only SCCP firmware.
Am I right?
Sergio
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Carlos [EMAIL PROTECTED] lazily top-posted:
Has anyone got a response from this?
It was just spam. Forget it.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL
Hi,
I'm seeing a problem where if I place a call, then forcibly quit or turn off
the client the call stays active.
The frames counters stop so its apparent the client has gone away but the call
remains active.
Asterisk is CVS-HEAD 23-Jun-05
What is supposed to happen in this scenario?
Hi Guys
Just a quick question. Does * write directly into PGSQL
database like MySQL?
Kind Regards
Terry Wade
Mobile: +27 82 802-5750
Office: +27 11 784-7642
Fax: +27 11
388-0855
Linux is
like a Wigwam - No gates, no windows, Apache inside
Disclaimer
and Confidentiality
Are there any other GDK users out there with Asterisk?
Ive got all the integration working, except
voicemail.
Does anybody know a way of disabling the forward to
voicemail on a per extension or per DDI basis (I can disable the voicemail hunt
group but then I cant light the MWI
Kristof Hardy [EMAIL PROTECTED] wrote:
Matt Brooks wrote:
I am just emailing to inform you guys that a new website has been
created for asterisk.org. You can find the beta site up at
http://beta.asterisk.org. It utilizes the drupal portal framework and
Looking very good and much easier
Hi,
Has anyone manage to comile app_dbodbc or ast_data with the latest
stable release (1.09). If so can you give some guidence on howto do it
as I have trouble getting either working.
Umar
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Asterisk-Users mailing list
Hello
I have an application for Asterisk which could
involve potentially 5000 or more extensions. Possibly this number of
people making calls. All calls would be internal. Could enough
hardware be thrown at the problem to make this work? Anyone setup an
installation of this size? Any
I just checked my account via https://secure.inetm.net and my balance is
visible where it always has been on the billing activity page.
*shrug*
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chad Brown
Sent: August 3, 2005 01:55
To: Erik
Hello
A lot of my customers have people who are in the
office most of the time but occasionally wish to work from home. So they
may have a sip phone which is extension 208 in the office. When they work
from home they can of course plug in a sip phone into their broadband connection
and
Hello,
routing based on DNIS is dependant on what your telco sends you. Usually on
Robbed-bit T1s(RBS) they will send you ANI and DNIS together separated by
stars like this:
*7275551212*1234*
(where 7275551212 is the ANI[callerID] and 1234 is the DNIS[last 4 digits of
the number dialed])
In
Kevin Walsh wrote:
Carlos [EMAIL PROTECTED] lazily top-posted:
Has anyone got a response from this?
It was just spam. Forget it.
I have an account with them, just waiting for a suitable ATA to arrive.
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305)
I just wondered - might save me some development
effort!
Angus
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To UNSUBSCRIBE or update options visit:
On Tuesday 02 August 2005 16:50, [EMAIL PROTECTED] wrote:
I know that a 3GHz P4 box with 1GB ram, Intel 815 chipset can handle 120
... Excellent description of a specific benchmark snipped ...
Of course, I can't answer the question as to minimum CPU - I only have the
CPU that I have.
May I
On Tuesday 02 August 2005 14:35, Michael D Schelin wrote:
Rich is correct. Example: Night security guards may need to catch an
inbound calls that could ring at more than one station. Maybe one is
doing rounds and the other is at another desk off site. Sometimes call
forwarding is too slow.
asterisk-users
How to config incall ?I have a E400p card
but How to config incall ?
thanks a lot.
E400P - Quad Span E1 Card
outcall can set:
# more extensions.conf
[default]
include = from-sip
[from-sip]
exten = 200,1,Dial(Zap/1);
exten =
All of these postings about ringing two (or more) phones is well known
and fairly well understood by everyone. The issue that everyone seems
to want to ignore in the postings is the busy lamp field functionality
of key systems (not pbx's). I'm not the OP and I've been around *
and sip phones for
Wilson Pickett wrote:
I have IAX2 (FWD) partially working. I can place calls from my
Asterisk box but I cam unable to receive them (comes back as
busy). I have my firewall forwarding the udp ports 5060, 4569,
5036 and 1 thru 2 to my asterisk server. I think I have
the firewall correctly
asterisk-users
E400P - Quad Span E1 Card
How to test E400p card without E1 lines?thanks a lot
May I loop the card?
how to do ?
dev2002
[EMAIL PROTECTED]
2005-08-03
Just a data point... tftp works just fine in RHv9 and FC3 with remote
7960's. Images, config files, etc, get transferred correctly every time,
and the 7960's are between elcheapo firewall boxes.
If you really want to restrict who can access the tftp server, run one
of the firewall app's on the
I have an ISDN card, Billion ISDN PCI Card
I tried to use the ZAPHFC, I patched the kernel, I did anything (also
followed reccomandation on use on Suse Linux Professional 9.2 --my box is)
using bristuff last version.
In the end I succesfully compile zaphfc, but I am not able to use the card
(a
For those of you who have been working with asterisk for a while and
who have experience with SIPxchange, why have you chosen Asterisk over
the latter?
What are some significant differences between the two that those of
you familiar with both have discovered?
Brent
It's not just him. The list was majorly down from sometime on the 29th
until the 1st.
MARK.
Derek Whitten wrote:
must be just you.. get messages all day every day here..
:-)
On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote:
This is usually a very active list, but looking
There is DEFINITELY 7970 SIP firmware out there...maybe Betabut it's
out there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 03, 2005 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Rich Adamson wrote:
Just a data point... tftp works just fine in RHv9 and FC3 with remote
7960's. Images, config files, etc, get transferred correctly every time,
and the 7960's are between elcheapo firewall boxes.
If you really want to restrict who can access the tftp server, run one
of the
A lot of my customers have people who are in the office most of the time but
occasionally
wish to work from home. So they may have a sip
phone which is extension 208 in the office. When they work from home they
can of course
plug in a sip phone into their broadband
connection and work
I ran across AstLinux today, and noticed they had a build for Soekris
Engineering net4801. Is anyone running this board with AstLinux in a production
environment? If so, what type of load have you been able to put on it? Any luck
getting Digium hardware to run on it?
Any other
Sounds to me like your phone vendor is talking out of his arse.
You should be able to place a crossover cable between your * box and
your pbx. They both think the other is a phone company.
I've done this with Avaya Definity G3's a few times now and it works fine.
Mark
Administrator TOOTAI
Hi Tim,
I would like to test it as well.
Thanks,
Derek
On Wed, 2005-08-03 at 00:37, Boris Zolotarev - Pamet wrote:
Hello Tim,
I am definitely interested in testing it.
Please contact me off the list.
Best Regards,
Boris.
If anyone is interested I'm (slowly) developing a
GPL'd
Hi Eric -
I am having trouble with one of our IP600. Every five days or
so, the phone locks up. This is the third 600 I have put in place. I
am running asterisk 1.0.9. Has anyone had this problem with the
IP600?
What version of the bootrom and sip firmware are you using? Can we
Tim Litwiller wrote:
I've been using * at home at my house for while and like it but for
work I didn't know the answers to these questions.
But now my new employer is wanting to upgrade a very old phone system
and wants to make sure our new system has some features
I've talked to him about
[EMAIL PROTECTED] ha scritto:
In the end I succesfully compile zaphfc, but I am not able to use the card
(a lot of problem running zapcfg, a loto of problem starting asterisk
saying about wrong anything (from signalling to any other parameter
specified in zapata.conf)
You may want to post
Hello
I want to setup an Asterisk with three analog
lines. Two of the analog lines are the main office number. The other
line is the fax number. The fax machine plugs into the line 3 but also
will be a connection to the third port on a Digium analog card.
Reason for the third line into
Yep, I second (or third) that observation.
Ryan
It's not just him. The list was majorly down from sometime on the 29th
until the 1st.
MARK.
Derek Whitten wrote:
must be just you.. get messages all day every day here..
:-)
On Mon, 2005-08-01 at 05:49, Howard Leadmon wrote:
This
Hello..
I am new to the asterisk/astcc domain and have to do some maintenance work
on an existing system. As far as I know astcc has been installed and has
worked previously. All of a sudden it has stopped working. Since I am not
aware of how the interfacing between astcc and asterisk, I need
This seems to be due to a driver conflict. If I unload Zaptel, the
sound returns.
I'm having the same issue with a 2.4 kernel on whitebox 3 using HEAD.
Still investigating... let me know if you find anything new.
Jack
On 6/29/05, Jeremy McDermond [EMAIL PROTECTED] wrote:
I've looked all
I'm having the same issue. If I unload Zaptel, and restart
asterisk... the sound does return.
On 7/25/05, Arnd Vehling [EMAIL PROTECTED] wrote:
Hi,
i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07
and everything worked fine sofar when suddenly the voicemail and
Finally got everything up and run with the help of Manny
Wise last night. So I am setting up my digital assistant and getting down to
the task I need this box to perform the most. I need to have a custom app that
I can call that will take me pressing 2 at the menu and have it transfer the
Rich Adamson wrote:
All of these postings about ringing two (or more) phones is well known
and fairly well understood by everyone. The issue that everyone seems
to want to ignore in the postings is the busy lamp field functionality
of key systems (not pbx's). I'm not the OP and I've been around
Hi all,
I'm thinking of setting up an Asterisk based VoIP system between two offices
and I wanted to know if it is possible to intercept calls with Asterisk if
so how does one set it up?
Thanks.
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Yep, another list posting on this topic :)
All the messages I've read on this are from people experiencing these errors
in quiet times - I get them as soon as I plug a port on our TE410P to an
Inter-Tel AXXESS PBX.. and I get them continuously...
I'm just sticking an * box in between ISDN30e
Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Kevin Walsh wrote:
Carlos [EMAIL PROTECTED] lazily top-posted:
Has anyone got a response from this?
It was just spam. Forget it.
I have an account with them, just waiting for a suitable ATA to arrive.
Good for you. Personally, I never
Hi all,
Im having problems using a fax machine conected trough a grandstream
286 sip ATA, it must be able to send and recive fax from pstn, but fax
always ends with communication errors 252/244/232 and others.
Im using alaw/ulaw codes on pass trough mode, also have tried asterisk
faxdetection,
On Wed, 2005-08-03 at 07:52 +0200, Wilson Pickett wrote:
In the example below if I dial valid extension 1000, the Invalid
context plays pbx-invalid as it is included with _7 context.
Include voicemail in the main context.
Thanks, I new it must be something simple.
Simply reposition the
Hi
I am ordering a ISDN 30 line in from BT to use with digium hardware.
Was wondering if there was anything specific I should ask for when getting the
service in place.
Thanks
Start your day with Yahoo! - make it your home
app_dbodbc has been publically deprecated by the author and he isn't updating
it. Functionality provided by ast_data is provided by RealTime. You will need
CVS-HEAD to use RealTime. Or wait a month for 1.2 to come out.
-Matthew
Quoting Umar Sear [EMAIL PROTECTED]:
Hi,
Has anyone manage to
Has anyone tried this? I got in to download but now I can not get back
into mozdev.org. It did not come with any directions or help. If anyone
has it working where did you get instructions?
TIA
Bob
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Buy a 3 porst fxo card and 1port fxs (green) card from digium.
Plug your fax the the fxs port.
Assign an extension to the fax at extension.conf
Create a menu.
Since the call will be bridged from fxo to fxs natively, there is very
few loss and the fax works ok.
Anyway, the diferrence between
On Wed, 3 Aug 2005, Emanuele Pucciarelli wrote:
is it possibile to do not use zaphfc and configure in some way a
CHAN_CAPI
channel pointing to Billion card ??
I don't think so, unless someone has written a CAPI layer for HFC-S PCI A
cards!
Isn't mISDN providing this?
Armin
John Novack wrote:
Tim Litwiller wrote:
I've been using * at home at my house for while and like it but for
work I didn't know the answers to these questions.
But now my new employer is wanting to upgrade a very old phone system
and wants to make sure our new system has some features
Thank you for your answer.
anyway I just destryed my linux box, and I am installing it again.
The problem was, I think, that the driver was not loaded, sayng something
about pci card not found.
Really funny, becouse the Yast detected it and let you configure it.
In the end all the modules
Hello,
I start trying to use a USB dongle and a Bluetooth GSM phone to make GSM
call with asterisk using the BLT channel provided by the GSM phone.
Unfortunately I get a Everyone is busy/congested at this time whenever
I try to Dial(IAX2/[EMAIL PROTECTED]/2, BLT/MotorolaLara/3474501***)
For
Ok I've got SCCP running I have my 7970 firmware files.
Can anyone send an XMLdefault config and an SEP config file?
There are a bunch of sbn files in the package...not sure what needs to
be loaded.
-Darren
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I think what you want is called DISA
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DISA
DISA (Direct Inward System Access) Allows someone from outside the
telephone switch (PBX) to obtain an internal system dialtone
and to place calls from it as if they were placing a call from
I assume you mean to save on tolls between the two offices.
If so, the simplest way is to set up asterisk on both ends
and specify in your dialplan which numbers you want to
go out over IP and which you want to go out over landline.
Asterisk makes this easy as it uses the most specific first,
so
http://www.digium.com/index.php?menu=compatibility
What servers does one recommend
though using ? Our company hates using HP junk, dell used to be a good choice
for most of our stuff. IBM is way overpriced. Anyone have any suggestions?
Sascha
From: [EMAIL PROTECTED]
Angus Comber wrote:
I just wondered - might save me some development effort!
Angus
http://www.gnomemeeting.org/ ?
Jorge
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To
Supermicro's can be nice. Problem is that Supermicro's aren't sold in Canada
and as per our specification is it needs to be a tower based server.
Anyone know any other decent fully manufactured systems? .. Systems with
support on them like dell, that would work well with asterisk.
S.
Take a look at http://moziax.mozdev.org/
Take care.
On Wed, 2005-08-03 at 11:11 -0400, Robert A. Rawlinson wrote:
Has anyone tried this? I got in to download but now I can not get back
into mozdev.org. It did not come with any directions or help. If anyone
has it working where did you get
On Wed, 2005-08-03 at 10:23 +, Andres Tello Abrego wrote:
Assign an extension to the fax at extension.conf
Create a menu.
Why even bother to do that much? Just put the 3rd port/line into its own
extension where s automatically dials the fax machine on 4. You can
still use 1, 2, and 3 for
Robert A. Rawlinson a écrit :
Has anyone tried this? I got in to download but now I can not get back
into mozdev.org. It did not come with any directions or help. If anyone
has it working where did you get instructions?
The project home page is:
http://moziax.mozdev.org/
On Wednesday 03 August 2005 17:33, Jens von Bülow wrote:
Gavin,
Any ideas/advice would be warmly received right now!
You are not going to like my response...
Erk :)
The only way I could get this to work (luckily I had 2 identical sites and
was busy with the upgrade to the gen2 card) was
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Why do you put me down? I have not done a thing to you and I'm not a
spammer. Please stop this activity It's not professional. If I were to
give you bad service please feel free to comment negatively but I've
never dealt with you nor do you have an account with us.
Sincerely
Michael D.
Hi,
Sorry for the top post, but the precedent has already been set. :-(
We're using a Dell 1850 with a TDM04B without any problem so the
previous
post is incorrect about TDM cards not working in this machine. We're
using
it with Fedora Core 1. The only problem was that we had to add a
PCI
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