I want to interconnect 2 pbx switches from to
distinct location via an internet vpn using asterisk as VoiP
gateways.
The problem is what interfaces i must use between
asterisk servers and pbx switch (FXO or FXS), and why?
thank you in advance
___
Altus Snyman wrote:
Where do you get this new version of bristuff,I had a look on the
webpage and there's only RC3
My first action every morning is to look at the top of this page:
http://www.junghanns.net/asterisk/downloads/?C=M;O=D
--
Best regards
Peer Oliver Schmidt
Michiel van Baak wrote:
On 05:14, Tue 08 Feb 05, Mazhar Hussain wrote:
If this sounds usefull to you, reply so on the list and I
will try to setup a clear txt doc where and how to find the
sourcecode.
I would like to see the information you can provide on this.
Thanks,
Matt
--
Matt Gibson
VOIP
Hi Michael,
-Original Message-
dIf you reread his email, he is stating that he has a quadbri
So do we. We are seeing something similar, even on RC5.
On Wed, 09 Feb 2005 07:58:38 +0100, Peer Oliver Schmidt
[EMAIL PROTECTED] wrote:
Altus Snyman wrote:
We have a quad bri
Hi
I am trying to get app_pppd to make an outgoing call to my ISP.
Has anybody got this to work yet?
Thanks
Roger
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I get the following error when trying to compile asterisk
1.05 on red hat 9.
[EMAIL PROTECTED] asterisk]# make install
*** You don't have mpg123 installed. You're going to need
***
*** it if you want
MusicOnHold
***
./mkdep -pipe -Wall -Wstrict-prototypes
-Wmissing-prototypes
On Wed, 9 Feb 2005, Stefan Gofferje wrote:
I wanted to try out app_icd but...
[EMAIL PROTECTED]:/opt/app_icd make
=== Compile: /opt/app_icd/app_icd.c (app_icd.o)
app_icd.c: In function `app_icd__log_events':
app_icd.c:2104: error: structure has no member named `cid'
app_icd.c:2104:
I'm not sure it answers all your questions but there is ast-stats from
http://areski.net/areski/index.php?
option=com_contenttask=categorysectionid=5id=70Itemid=54
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On Tue, 8 Feb 2005 14:27:28 -0500, Brian C. Fertig
[EMAIL PROTECTED] wrote:
Does anyone know how to get rid of these hung channels?
I am getting this when I do a:
show sip channels
209.82.xxx.xxx0071495217 2591218534@ 00103/1 unknow(d)
209.82.xxx.xxx0041590104
Hello, everyone
After having spent several time to look for any
solution for my problem, I decided to write here.
Here is the problem got a Digium X101 FXO card and
Digium TDM400P alias freshmaker (1 fxs module on it on
first port ) in my asterisk box. The X101 works
perfectly.
The problem
Hello,
I am attempting to use Asterisk as a protocol converter.
I have set up asterisk to route incoming h323 calls to a SIP termination
carrier.
I make a test, call is coming correctly, is rerouted to termination carrier.
Call connects and phone rings. Then, I pick up the phone and it
Our SIPS implementation is absolutely standard conform according to RFC3261
and our SRTP implementation follows RFC3711.
Regards
Nils Ohlmeier
On Tuesday 08 February 2005 13:37, Remco Barende wrote:
What about SIPS (Secure SIP)?
I cannot find anything about it in the Wiki but the Snom
Yesterday I setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds terrible - clipping, buzzing,
digital distortion, and its too loud (which probably isn't helping) and
I'm just running it thru the 'default' line in music onhold.conf line
default =
David Brodbeck wrote:
-Original Message-
From: David Brodbeck [mailto:[EMAIL PROTECTED]
Okay, the problem appears to be that I'm tone deaf. ;)
I finally thought to turn on debugging on the channel. The
PBX is sending
D, not *. The programmer of the previous voice mail system (whose
When attempting to call one of the example numbers, like 17474745000, I
only get 488 Not Acceptable Here. It works fine when I configure the
softphone (Xten X-Lite) to use sipphone's server directly. Am I missing
something? Here's my relevant config sections:
sip.conf:
in [general]:
register =
On Wed, 2005-02-09 at 10:20 +0200, Roger Wrethman wrote:
Hi
I am trying to get app_pppd to make an outgoing call to my ISP.
Has anybody got this to work yet?
Any reason you can't use a .call file to initiate the call? And just a
simple reminder, it has to be ISDN.
--
Steven Critchfield
I just discontinued my BV service, however CID was working just fine from
a 303 line on cvs head. If you post the relavent sections of exentensions.conf
and sip.conf, we might be able to suggest a couple of things.
Outbound callerid via BV to the pstn will not show anything more then
the number
hello
any one using cvs version of asterisk(realtime
addons). i have defined two users 2000 and 3000 in
sip.conf. after that when i try to call 2000 from 3000
or try to call 3000 from 2000 it is giving me 404 Not
Found error.
Found user '2000'
Looking for 3000 in default
Reliably Transmitting
After having spent several time to look for any
solution for my problem, I decided to write here.
Here is the problem got a Digium X101 FXO card and
Digium TDM400P alias freshmaker (1 fxs module on it on
first port ) in my asterisk box. The X101 works
perfectly.
The problem comes from
Good day all
We have 2 asterisk servers,connected with iax2 and the phone via SIP
They dont have a very big line so I want to restrict the call limet to 3
iax2 calls at a time,and for instance it the 4th call is made it will
say something like all lines are being use try later
Please help
thanks
Hi there
I am new to Asterisk and am looking for a web based management tool, for
managers to manage hunt groups, extensions etc and for user to have access
to there own phone features. I have seen there are a number of commercial
tools available for this, but I presume there are some freeware
I've got one freaky budgetone that wont work using dhcp assign ip
address via mac code.
Basically I need to assign it an ip address using the phones internal
web server.
Maybe this was your problem as well.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
hello
any one using cvs version of asterisk with realtime
mysql addons. i am having a problem with it. i have
defined two users 3000 and 2000. when i try to call
3000 from 2000 it is giving me '404 Not Found' and
saying Found user '2000' and Looking for '3000'
but when i try to call 2000 from
David J Carter wrote:
How do you want Switch to appear to Asterisk.
1. As an extension. Then use an FXS connection to a CO line input.
The extension interface at the PBX will be supplying battery and dial
tone. Therefore, you would want to use the FXO (red) daughter board on
your TDM400P card.
I want to interconnect 2 pbx switches from to distinct location via an
internet vpn using asterisk as VoiP gateways.
The problem is what interfaces i must use between asterisk servers and pbx
switch (FXO or FXS), and why?
You must use FXS ports on *, then plug these in you PBX as phone
I've discovered that one of the pitfalls of wanting to try out the new
jitter buffer is that you have to move to CVS head... Which isn't a
biggie unless you've been using mysql without odbc. Am I dreaming or is
the old type of non-odbc sql support eliminated from cvs head?
Anyhow, just thought
On Mon, Feb 07, 2005 at 11:16:05PM -0600, Eric Rees wrote:
Has anyone seen this message trying to install an TDM400.. spurious
8259A interrupt: IRQ7
Not sure what to has to do with your system, but I read somewhere that
it is related to how the original interrupt controllers worked. If a
card
OK, I've spent way more time than I wanted to on getting
an x100p clone to work in Australia. I'm happy to consider
other (more functional) options.
Does anyone have an opinion on both the Sipura 3000 and
other Digium cards (like the TDM400P)?
I need something that works with no much
Mark,
I have heard this problem. I'm not exactly sure what the cause is but
check for any duplex mismatches between the phone and the * box.
Hope this helps.
Scott H
Mark Benson wrote:
Yesterday I setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds
We are having problems with DTMF generation with our supplier of IP to
PSTN call termination. Their (Entice) soft switch is looking for RFC2833
payload type of 99 but Asterisk is using RFC2833 payload type 101.
We are specifically having problems being able to access IVR menus and
voice-mail.
Hi
I am running mediaproxy infront of asterisk, with SER
xlite SER ===asterisk (voicemail)
||
||
||
Mediaproxy
If xlite is behind NAT or not, the mediaproxy replaces the c= header in
the SDP part with th IP address of the mediaproxy (tks
On Tue, 8 Feb 2005 22:09:59 -0800 (PST), Paul Chan
[EMAIL PROTECTED] wrote:
Hi All,
I have a question about Fastagi because I can't get
it to work for some reason. Everytime I execute the
fastagi command, i get an error:
my extensions.conf:
..
exten = 1000,1,agi(agi://some_ip_address)
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
--
#Joseph
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Joseph wrote:
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
I got fed up with the yo-yo, which then led me to dump fwd and install
asterisk and start playing with inter-asterisk routing via e164.org...
--
Best regards,
Duane
http://www.cacert.org - Free
I installed mpg123.0.59s and that was nasty so installed 0.59r but it
was still distorted, eventually deleted s and reinstalled r and after a
few mins the music on hold sorted itself out - it just happend as I was
testing it after reinstalling - weird - I had looked at the
phone/asterisk
No problems here - works fine.
- Original Message -
From: Joseph [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 09, 2005 8:17 AM
Subject: [Asterisk-Users] IAX = FWD down again?
Can anybody confirm
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
Works fine for me.
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I get the following error when trying to compile asterisk 1.05 on red
hat 9.
Is this the tarball available for download from the asterisk website?
You might try CVS instead - try the CVS HEAD release:
# cd /usr/src
# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
# cvs login
Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download
at sourceforge and does exactly what you are looking for.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brett,
Gary
Sent: Wednesday, February 09, 2005 8:01 AM
After having spent several time to look for any
solution for my problem, I decided to write here.
Here is the problem got a Digium X101 FXO card and
Digium TDM400P alias freshmaker (1 fxs module on it
on
first port ) in my asterisk box. The X101 works
perfectly.
The problem comes from
I have both Stable version (asterisk-1.0-RC2) and the CVS version (asterisk
v1-0-5) running on different Red Hat 9 boxes and there is no problem. I have
only problem when I installed the oh323 driver (asterisk-oh323).
Make sure you install Red Hat with required Package to run Asterisk.
Regards.
How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ?
Regards.
Daniel.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins
Sent: mercredi 9 février 2005 15:42
To: Asterisk Users Mailing List - Non-Commercial
I just want one of my incoming numbers to go to an IVR service that will
allow me to select what I want.
For example
Press 1 for Mike, 2 for Karen, 3 for other, 9 for voicemail etc
Just need to learn how to configure services now so that I can put a menu
on
one of my numbers!
Elaborate
That would be the AMP database, I don't know.
Ping the amp list and find out.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa
Sent: Wednesday, February 09, 2005 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello all,
Is X-lite soft phone support G.729? I actually
use it but there is no G.729 support. Anyone know where to have it?
Regards.
Daniel.
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I try to use meetme app
after reading manual i compile and install zaptel with ztdummy
when i make lsmod
i have
ztdummy 2532 0 (unused)
wcusb 20064 0 (unused)
zaptel179168 4 [ztdummy wcusb]
usb-uhci 26348 0 [ztdummy]
hello,
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
Anybody could help me to configure Asterisk in order
to set instant message and presence ?
I've tried with Ondo sip server it's ok !
Regards
I regularly get asked by business people, What's the point of VoIP?,
so I put together a guide:
http://integrics.com/tips/voip_for_business/
I'd be interested in hearing your feedback, and ideas for expansion.
--
Alistair Cunningham,
Integrics Ltd,
Telephony, database, Unix consulting worldwide
Joseph wrote:
When I try to load iax.conf I get (*-1.0.5):
loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/iax.conf:
cannot open shared object file: No such file or directory
iax.conf is not something you can load. chan_iax2.so is, though.
___
Altus Snyman wrote:
Good day all
What is the file sip_notify.conf for
Read the Mantis bugnotes about it when it was added. It's very useful.
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Altus Snyman wrote:
Good day all
We have 2 asterisk servers,connected with iax2 and the phone via SIP
They dont have a very big line so I want to restrict the call limet to 3
iax2 calls at a time,and for instance it the 4th call is made it will
say something like all lines are being use try later
Hi Harry -
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
What version of the SIP firmware are you using? I've had success with
1.3.0, 1.3.1, 1.3.4, and 1.4.1.
My sip.conf entries for my Polycom phones look like this:
[12]
All,
I followed the channels/h323/README to the letter and everything does
compile properly. When I start asterisk I get the following error:
[chan_h323.so]Feb 9 10:30:51 WARNING[30700]: loader.c:301 __load_resource:
libpt_linux_x86_r.so.1.8.1: cannot open shared object file: No such file or
Greetings,
2 nights ago I upgraded one of my remote servers to the
latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and
outbound caller ID stopped working. At first I thought it was my carrier, so Ive
been yelling at them for the past day or so.
I then upgraded
Are you running the latest stable? I think there's a bug with it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Sunday, February 06, 2005 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
Paul Belanger [EMAIL PROTECTED] wrote:
[...]
My question is, how come the LD_LIBRARY_PATH defined in /etc/profile
did not link the libs properly?
LD_LIBRARY_PATH is occasionally ignored for security reasons.
If you wish to globally add a directory to the library search path,
you should put it
you have to buy pro version for 729, $50.
-Matthew
- Original Message -
From: Daniel Eboa [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 09, 2005 9:00 AM
Subject: [Asterisk-Users] G.729 codec for
-Original Message-
From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED]
I'm prbably stupid, but wont this do what you want?
exten = 1,1,Goto(bye,s,1)
No, because I wanted to match on D, not 1.
Anyway, I figured it out. The extension was working, but Background()
ignores the tones A
On Thu, 2005-02-10 at 01:22 +1100, Duane wrote:
Joseph wrote:
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
I got fed up with the yo-yo, which then led me to dump fwd and install
asterisk and start playing with inter-asterisk routing via e164.org...
Michael Graves wrote:
Hi All,
I was just reading through Info Week while on a flight and happened
upon an brief piece about a new VOIP security intiative worked up by a
handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All
of this begs the question of can't we get just do this as a
Michael Welter wrote:
SER newbie here. Why do you need Asterisk for Sip-SIP setup? And if
there is a reinvite, is that for the RTP stream only or for the SIP
transactions as well? Will you lose the BYE transaction if there is a
reinvite?
Also, how many SIP registrations do you expect to
Why not let asterisk be your PSTN GW? It is in our case, just throwing out
my $0.02.
Most of the cases I can think of I can get around. The one I can't seem to
figure out is 'Agents'.
Agents will need to login/logout using 1 number. I can forward that number
from SER to asterisk by looking for
David Brodbeck wrote:
Anyway, I figured it out. The extension was working, but Background()
ignores the tones A through D by default. I didn't realize this because I
wasn't waiting for message playback to finish.
Please enter a bug in Mantis for this; it should very likely be
corrected, as I
[EMAIL PROTECTED] wrote:
I think you might be missing the point here. SER is a raw SIP processor.
So for a second throw everything you know about Asterisk + SIP out the
window and go back to vanilla SIP. Getting used to a B2BUA in the call
path kinda beats some of the raw power of SIP
Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution
for small systems?? I am looking for an open source web management tool to
use on any size asterisk server (even ones that are already up and running)
the user base could be anything between small and large with many
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
-Brett
Aw crap. I completly forgot about call pickup. Good point. If you have a
call come into one of your
Norman == Norman Howlett [EMAIL PROTECTED] writes:
Norman We are having problems with DTMF generation with our supplier
Norman of IP to PSTN call termination. Their (Entice) soft switch is
Norman looking for RFC2833 payload type of 99 but Asterisk is using
Norman RFC2833 payload type 101.
From
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
David Brodbeck wrote:
Anyway, I figured it out. The extension was working, but
Background()
ignores the tones A through D by default. I didn't realize
this because I
wasn't waiting for message playback to
Hi All,
I just installed Asterisk 1.0.5, and the
installation went fine (I ran modprobe zaptel and
modprobe wcfxo). However, when I ran ztcfg I get the
following:
ioctl(ZT_LOADZONE) failed: Invalid argument
Notice: Configuration file is /etc/zaptel.conf
line 135: Unable to register tone zone
Matthew Boehm wrote:
If I can get re-invites working great, then I should have no worries about
inter-office communication. SER should be able to connect 2 office-mates to
eachother even if they are both behind the same NAT, or behind different
NATs.
You can accomplish that with a low-end box
On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote:
David Brodbeck wrote:
Anyway, I figured it out. The extension was working, but Background()
ignores the tones A through D by default. I didn't realize this because I
wasn't waiting for message playback to finish.
Please enter
Matthew Boehm wrote:
Why not let asterisk be your PSTN GW? It is in our case, just throwing out
my $0.02.
Most of the cases I can think of I can get around. The one I can't seem to
figure out is 'Agents'.
Agents will need to login/logout using 1 number. I can forward that number
from SER to
Matthew Boehm wrote:
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
-Brett
Aw crap. I completly forgot about call pickup. Good point. If you have a
call
[snip]
It is working again :-)
It appears something broke after recent upgrades (on Gentoo) as I wasn't
even able to dial VOIPJET, though I don't know what was broken :-/
I re-emerged asterisk and it is working gain.
--
#Joseph
I wander what is causing the problem, I was thinking that it was
Here is the situation at hand, awaiting input from the collective minds
of the asterisk-users list:
1. I have Asterisk registered as a SIP user agent to another Asterisk
(which stands for the real provider, to another location).
register = user:[EMAIL PROTECTED]
2. I have a peer section for
I'll ask a stupid question, how does a user hit an alpha letter from his
touchtone?
I know that the Cisco 7960's support entering alpha letters, and it could
potentially do it (maybe), but how does the average end user enter an a b c
or d from their touchtone phone?
-Original Message-
Steven Critchfield wrote:
I would suggest that that be added as a optional switch to background to
get the extra digits. While I do not know if A-D are easy to hit with
Talk-Off, it is 4 more potential digits to hit. Also it would be less
surprise to a user to be required a flag to Background() to
David Brodbeck wrote:
-Original Message-
From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED]
I'm prbably stupid, but wont this do what you want?
exten = 1,1,Goto(bye,s,1)
No, because I wanted to match on D, not 1.
I am stupid - I thought you meant the DTMF for the D button (aka 3DEF) :-)
[EMAIL PROTECTED] wrote:
I think most people probably do something like:
AddQueueMember(techsupport|SIP/${CALLERIDNUM}), but I bet you can put
any valid channel name in there.
And you would win that bet :-)
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I search a ActiveX to develop one softphone SIP with codec G723. Who can
help me?
Thank´s
João Carlos Moura
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You need to go back and reread.
It is just pretty much an asterisk configuration tool (ok some minor things in
the backend but it's the best out there).
AMP is available for free download but they make their money by offering
support.
[EMAIL PROTECTED] is just an automated way of installing
I know FWD uses a SER/Asterisk combo, and I keep hearing
about the massive benefits, however, my initial playing around in SERs
configuration indicates its NOTHING like Asterisk at all, and almost 5x
as difficult to understand and configure. But thats only after a few
hours of playing with
Paul Rodan wrote:
I'll ask a stupid question, how does a user hit an alpha letter from his
touchtone?
I know that the Cisco 7960's support entering alpha letters, and it could
potentially do it (maybe), but how does the average end user enter an a b c
or d from their touchtone phone?
Some phones
-Original Message-
From: Steven Critchfield [mailto:[EMAIL PROTECTED]
On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote:
David Brodbeck wrote:
Anyway, I figured it out. The extension was working, but
Background()
ignores the tones A through D by default. I didn't
Paul Rodan wrote:
I know that the Cisco 7960's support entering alpha letters, and it could
potentially do it (maybe), but how does the average end user enter an a b c
or d from their touchtone phone?
They don't. Most phones (99.9%) don't have any way to generate DTMF A
through D. There are test
On Wed, 9 Feb 2005, Matthew Boehm wrote:
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
Aw crap. I completly forgot about call pickup. Good point. If
Hello,
2 nights ago I upgraded one of my remote servers to the latest CVS Stable,
Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID
stopped working.
My suggestion would be to downgrade to 1.0.3. It might solve your
problem. There were a number of changes in callerid
Gary,
contact me off-list. I have developed a GUI Windows based tool that will
allow
management of configuration files if you are running RealTime. It supports
sip,iax,extensions,voicemail currently. It will also display CDR's and the
various
schema's used by the Asterisk box.
Tom Chandler
Peter Svensson wrote:
On Wed, 9 Feb 2005, Matthew Boehm wrote:
I'm still not sure how to provide services that interact one phone with
another phone's RTP stream. Like call pickup. How can I pickup a call on
another asterisk server? Hmm Hm
Aw crap. I completly forgot about call
On Wednesday 09 February 2005 12:53 am, Peer Oliver Schmidt wrote:
Steven Critchfield wrote:
astfax allows you to create an email to fax gateway.
Are we going to see some integration of astfax with Courier-MTA/IMAP?
If you look at the instructions, you only need to make a some form of
-Original Message-
From: Paul Rodan [mailto:[EMAIL PROTECTED]
I'll ask a stupid question, how does a user hit an alpha
letter from his touchtone?
I know that the Cisco 7960's support entering alpha letters,
and it could
potentially do it (maybe), but how does the average end
Thanks Dean, you say that [EMAIL PROTECTED] is just an automated way of
installing AMP and FOP and Web Meetme. But from the installation
instructions I have read, you download an ISO image that installs a linux
distro for you (destroying current install) and then configures itself for
use
I do
Hi list!
Sorry for this slightly off-topic message but does anybody know if the
standard for ISDN BRI is the same in Spain as it is in the rest of Europe
(or the Netherlands).
Will a standard HFC-S card work?
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On Tuesday 08 February 2005 5:42 pm, Remco Barende wrote:
Looks really cool :)
My company requires that for every fax we send we get a printed status
report that includes the number we sent the fax to, the number the other
fax reported, time+date, tx time and if the fax was sent ok or not
Joseph wrote:
I wander what is causing the problem, I was thinking that it was
something on my part but I did not change any settings and IAX2 registry
At the time the only thing I could put it down to was congestion...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote:
Oh right.. I remember seeing that.. yeah that looked a whole lot more
elegant than *8. Why isn't it in HEAD?
I'm not sure. Once it started getting some testing BKW closed it. If
someone is interested in testing the patch I'm sure the bug could be
I thought that as long as I stuck to the stable branch, only major bug fixes
would be included, no new features or changing of the way things are
handled?
I mean, isn't the latest CVS Stable better than 1.03? I'm in the
asterisk-cvs list and every day I see bug fixes added to the stable branch
Hello
all,
Is X-lite soft phone support
G.729? I actually use it but there is no G.729 support. Anyone know where
to have it?
Regards.
Daniel.
/Snip/
Daniel,
You know that X-lite
does not support G.729 and you also know where to have it, dont
you?
if you read your
questions a
I am running * 1.0.5 and have been having lots of problems with
outgoing calls and their sound quality. I am using ULAW for the codec
and sixtel for termination. Basically the problem is that portions of
the call seem to be lost and replaced with silence. Sometimes I can't
hear the person talking
Why not share with the community?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler
Sent: Wednesday, February 09, 2005 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Web based Asterisk
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