On Tuesday 18 September 2007 00:51:47 Kevin Kiely wrote:
Per the earlier recommendation, I picked up one of the ATS X10001P to
evaluate. I was able to configure the LAN for access, however, I don't see
where to enter the sip credentials. I have accessed the web interface with
root/test and
s/Trixbox/FreePBX/g
Please, Trixbox is a distro, the GUI is FreePBX.
Another option might be Destar. Google it up.
On 9/18/07, Matt Riddell [EMAIL PROTECTED] wrote:
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Ken D'Ambrosio wrote:
Are there any Asterisk GUIs out there that actually
Dear folks,
I'm using * realtime with no problems on most of the systems i've setup but
rarely i confront this problem that the asterisk doesn't load from database
when the systems rebooted and i have to reload it manually or restart it, but
it would work fine afterward, no problem how many
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Diego Iastrubni wrote:
s/Trixbox/FreePBX/g
Please, Trixbox is a distro, the GUI is FreePBX.
Except we were comparing with AsteriskNow - http://www.asterisknow.com/
(a distro) rather than AsteriskGUI -
Hi
On Tue, Sep 18, 2007 at 02:35:14PM +1200, Matt Riddell wrote:
I actually put my extensions.conf stuff into a generate.php file which
writes out the extensions.conf file with parameters supplied by the
customer stored in separate files.
So our software is doing the same thing
Hi!
I have a very strange question. I'm using trixbox with Asterisk
1.2.23-BRIstuffed-0.3.0-PRE-1y-j.
I configured and installed the HFC ISDN card with a script, as here:
http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox
Now i have 6 SIP hardphone, and softphone,
On Tue, Sep 18, 2007 at 10:20:14AM +0200, Péter Tóth wrote:
Hi!
I have a very strange question. I'm using trixbox with Asterisk
1.2.23-BRIstuffed-0.3.0-PRE-1y-j.
I configured and installed the HFC ISDN card with a script, as here:
What do you mean on direct call?
The error is more frequently on my sip trunk. Should I make a sip debug?
My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem?
Anyway i will watch the bri debug, and try to make a wrong and a correct
call.
Thanks
2007/9/18, Tzafrir Cohen
On Tue, Sep 18, 2007 at 12:07:20PM +0200, Péter Tóth wrote:
What do you mean on direct call?
The error is more frequently on my sip trunk. Should I make a sip debug?
My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem?
Anyway i will watch the bri debug, and try to make
Sorry if this comes thru twice, I had the wrong account selected to send the
first time...
Callers to the number get ringing, I get stuff in my asterisk console, and
it calls my softphone and ata, but answering either gets silence, and the
caller gets the ringing stop, if they wait ages they get
Hi!
Yes, the echo test worked perfectly.
When i try ztmonitor as follows, it gives strange output...
[EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv
Visual Audio Levels.
Use zapata.conf file to adjust the gains if needed.
( # = Audio Level * = Max Audio Hit )
My Asterisk installation crashes frequently.
Since it's a random event I am not able to reproduce
it so I can't say what is making it crash.
Here's a snippet of /var/log/asterisk/full just when
it crashes:
Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo
cancellation on channel 31
Sep 18
On Tue, 2007-09-18 at 05:15 -0700, Vieri wrote:
I have core dumps in /tmp.
What can I do to isolate the cause of these
segmentation faults?
You'd have to get an Asterisk developer to look at the backtraces
generated from those core files. There's more information on the
backtraces either in
On Tuesday 18 September 2007 15:15:38 Vieri wrote:
My Asterisk installation crashes frequently.
Since it's a random event I am not able to reproduce
it so I can't say what is making it crash.
Here's a snippet of /var/log/asterisk/full just when
it crashes:
Sep 18 13:42:51 DEBUG[378]
I thought this would interest a few people on the list - asterisk
enabled home security video recording dvr anyone?
http://deancollinsblog.blogspot.com/2007/09/bug-labs-opensource-hardware
.html
I had a really interesting conference call today about a new startup
called
Thanks for your replies, but the file you mention (cdr_mysql.conf) I have it
already configured since I'm already storing the CDRs in a MySQL database.
As I understand this file (cdr_mysql) is only for enabling mysql cdr storage
and cdr.conf should be used for cdr backend parameterizations.
Or
On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote:
stuff useless. My real concern was the immediate '/ignore' for asking
about an issue with the *now ditro that actually had nothing to do with
the GUI itself. Truth be told, most of my time today was in the CLI
You may be taking what happened
the GUI does NOT have meta-data in the sense you think of.
I can seamlessly operate a full PBX through the GUI and add
things myself, it is VERY simple.
The [default] context is global for the other numberplans (trunks/users)
There is no problem with overwriting files if you do it right.
When
On 9/17/07, Luís Palma [EMAIL PROTECTED] wrote:
Is there a way to enable the usage of UNIQUEID CDR field using a MySQL
database backend for storing CDRs without having to recompile
asterisk-addons as stated here
http://www.voip-info.org/wiki-Asterisk+cdr+mysql ?
After version 1.4 it is said
I know this borders on commercial, so I apologize. I will take this off list
as soon as possible.
Someone a couple months ago claimed to know how to get PRI or T1 voice
circuits significantly cheaper than going through the ILEC. I would
appreciate that person contacting me (off-list) at this
James FitzGibbon wrote:
On 9/17/07, *Jim Canfield* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
stuff useless. My real concern was the immediate '/ignore' for asking
about an issue with the *now ditro that actually had nothing to do
with
the GUI itself. Truth be told,
Also, update the asterisk GUI to what I have been working on now.
http://asterisknow.org/install-related to the asteriskNOW branch.
(This latest work includes VOIP Seamless service providers integration, and
also digital card detection and setup)
-bk
- Original Message -
From: Matt
James FitzGibbon wrote:
On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote:
stuff useless. My real concern was the immediate '/ignore' for asking
about an issue with the *now ditro that actually had nothing to do
with
the GUI itself. Truth be told, most of my
Hi List,
I have a client that has an interesting request. He wants to have people call
in to a conference room and all be able to talk however they should not hear
each other. There should be admin access to for one user to call in and be able
to listen in to everyone that is talking (they may
Does anyone know if the Dell Power Edge 1900 has an issue with
multiport E1 cards? We've had this server running for a while now with
2 E1 cards. At first we tried to install an Openvox D210P card with two
E1 ports but after a couple of kernel panics we thought that maybe the
card was
Here's what I'm showing in the logs...
[Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format
g729, extension(s) g729
[Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so = (Raw G729 data)
[Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module
version 32,
Quoting Dovid B [EMAIL PROTECTED]:
Hi List,
I have a client that has an interesting request. He wants to have
people call in to a conference room and all be able to talk however
they should not hear each other. There should be admin access to for
one user to call in and be able to
I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use
pri intense debug span 1
It is supposed to show every packet that is received on the PRI line.
I wanted to know in ISDN Pri when a call connects how are the data
(voice) packets (for PRI) shown in Asterisk. Or if there is
Carlos Chavez wrote:
Now the customer needs a third E1 port and since the computer pnly has
two PCI ports we decided to install a Digium TE411P card. Over the past
few days there have been several kernel panics. We have the latest
Asterisk 1.4.11, Zaptel 1.4.5.1 on CentOS 5 with all
Dovid B wrote:
I have a client that has an interesting request. He wants to have people call
in to a conference room and all be able to talk however they should not hear
each other. There should be admin access to for one user to call in and be
able to listen in to everyone that is talking
All,
I'm trying to configure queue agents w/ a DUNDi setup so that an agent
can login to whatever server they please w/o any custom setup. In
general this seems to work, agents login w/ AgentCallbackLogin into the
incoming context (not a special queue context) and can receive queue
calls.
The
Hi all,
Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command
pri intense debug span 1 , does it debug every packet received
(control and voice/data packets) ?
Thanks
--
Arpit Mehta
Graduate Student
Department of Computer Science
Columbia University
Tel:
Scott Moseman wrote:
Here's what I'm showing in the logs...
[Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format
g729, extension(s) g729
[Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so = (Raw G729 data)
[Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding
Thanks for the reply. I was not looking for a visualizer. I justed
wanted to see the data packets flowing in the asterisk CLI (for
example something similar to the rtp packets that flow when making a
voip call). I can see the various messages like CONNECT, SETUP etc.
I am a newbie regarding ISDN
Arpit Mehta wrote:
I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use
pri intense debug span 1
It is supposed to show every packet that is received on the PRI line.
I wanted to know in ISDN Pri when a call connects how are the data
(voice) packets (for PRI) shown in
On 9/18/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hi all,
Does Asterisk contain a full fledged ISDN packet sniffer. By giving the
command
pri intense debug span 1 , does it debug every packet received
(control and voice/data packets) ?
To get the equivalent of a packet sniffer, you'll
You cannot set up your dialplan with the CLI or am I missing something?
Creating relatively simple dialplans manually can be quite time consuming.
A GUI takes care of all that grunt work.
-Original Message-
From: SIP [mailto:[EMAIL PROTECTED]
Sent: Monday, September 17, 2007 2:12 PM
To:
On Tue, Sep 18, 2007 at 10:03:50AM -0700, shadowym wrote:
You cannot set up your dialplan with the CLI or am I missing something?
Creating relatively simple dialplans manually can be quite time consuming.
A GUI takes care of all that grunt work.
You write a dialplan with a text editor. Or
Arpit Mehta wrote:
Hi all,
Does Asterisk contain a full fledged ISDN packet sniffer. By giving the
command
pri intense debug span 1 , does it debug every packet received
(control and voice/data packets) ?
No, like I said in response to your other question, the only thing you
can
If the #AsteriskNOW channel is dead on IRC that does not mean you can
bring your problems to a channel dedicated to Asterisk (i.e. no GUI).
Go ahead and use AsteriskNOW, but don't pester the people in #asterisk,
most of which have never used it and many have never even heard of it.
All the
I don't think so, because in paging/intercom, the phones must support
Auto Answer.
The link you sent says:
SIP phones for the most part don't support any of these phone based
paging functions. If a SIP phone offers an Auto Answer function, you can
approximate limited paging intercom
Erik Anderson wrote:
On 9/18/07, Arpit Mehta [EMAIL PROTECTED] wrote:
Hi all,
Does Asterisk contain a full fledged ISDN packet sniffer. By giving the
command
pri intense debug span 1 , does it debug every packet received
(control and voice/data packets) ?
To get the equivalent of a
Arpit Mehta wrote:
Thanks for the reply. I was not looking for a visualizer. I justed
wanted to see the data packets flowing in the asterisk CLI (for
example something similar to the rtp packets that flow when making a
voip call). I can see the various messages like CONNECT, SETUP etc.
I am
What about using trixbox pro and forcing auto answer with the hud server
configuration?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED]
On 9/17/07, Dan Austin [EMAIL PROTECTED] wrote:
Lacy's response in the thread 'Why does
everyone seem to dislike *now?', has a small
bit that caught my eye.
Chan_Skinny made a lot of progress between 1.2 and
1.4, and even more in the later 1.4.X releases.
I am curious as to which
I am experiencing the exact same problem on solaris, and we do have
licenses purchased.
I will log a bug at digium in the next day or two about my particular
instance.
Scott Moseman wrote:
On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I hate to ask what may be a silly question,
On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I hate to ask what may be a silly question, but have you purchased
any G.729 licenses to use with the g.729 codec you downloaded?
If you haven't registered codec_g729 yet, that would be why you are
seeing this problem with codec_g729.
Zoiper can do it when you use the provisioning, contact me offlist on
[EMAIL PROTECTED]
Zoa
Joao Pereira wrote:
I don't think so, because in paging/intercom, the phones must support
Auto Answer.
The link you sent says:
SIP phones for the most part don't support any of these phone based
PSTN - g729 requires transcoding at that point.
You can however do:
G.729 phone - asterisk - G.729 phone without license (from my
understanding).
But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it
requires a license to preform transcoding.
--
Matt
-Original Message-
Does G.729 phone - asterisk - G.729 phone work with reinvite turned off?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson
Sent: Tuesday, September 18, 2007 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Fyi...
[myphone]
disallow=all
allow=g729
canreinvite=no
[otherphone]
disallow=all
allow=g729
canreinvite=no
I attempted this setup and it works. Media routed through the Asterisk.
Thanks,
Scott
On 9/18/07, Jeremy Mann [EMAIL PROTECTED] wrote:
Does G.729 phone - asterisk - G.729 phone work
The gateway is transcoding the PSTN into g729 and passing it to
Asterisk. The Asterisk never sees the PSTN from the outside. I have
watched the INVITE requests, they contain a request for a g729 only
call. But the INVITE to the phone does not include g729.
However, as previously stated, I did
Matt Watson wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman
My understanding was that it's not required for pass-through.
PSTN Phone - g729 Gateway - Asterisk - g729 Phone
Does this not equate to pass-through? Maybe I
Scott Moseman wrote:
The gateway is transcoding the PSTN into g729 and passing it to
Asterisk. The Asterisk never sees the PSTN from the outside. I have
watched the INVITE requests, they contain a request for a g729 only
call. But the INVITE to the phone does not include g729.
However, as
You have to increase the amount of available file descriptors per process:
http://hausheer.osola.com/docs/11%C2%A0%C2%A0
On Tue, 18 Sep 2007, Wai Wu wrote:
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk
Where do I get sound file for comfort noice. GSM or MP3 is fine.
Many thanks.
Jim
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asterisk-users
Follow me on this, it seems odd (or maybe I don't undertand)...
Test #1
[src_phone]
disallow=all
allow=g729
[dest_phone]
disallow=all
allow=g729
I can make the call (src to dest) and it will work using g729.
Both the call handling and media are going through Asterisk.
Test #2
[src_phone]
Thanks for the explanation and your cues.
I've been able to activate this feature by recompiling again asterisk-addon
source code (version 1.4.2).
If a runtime option is already undergoing in trunk, that's good news but for
now I prefer to stick to version 1.4.2. I'm trying to working with rpm
Thnx. That did it. I also reduce the stock size to 512K instead of 8M.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Tuesday, September 18, 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On 9/18/07, Wai Wu [EMAIL PROTECTED] wrote:
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build sip pvt data for asterisk1/700
Too many
On Tue, Sep 18, 2007 at 04:22:29PM -0400, Alex Balashov wrote:
On Tue, 18 Sep 2007, Wai Wu wrote:
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim
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Scott Moseman wrote:
However, in Test #3 the call will fail. Why?
Because Asterisk will attempt to use ulaw in preference to G.729 if
possible, and the other endpoint offered to support ulaw. The format(s)
supported by the eventual call destination are not relevant, because at
the time
Yes it is supported on cdr_mysql.conf.
I just have been looking to the example file (cdr_mysql.conf.sample) in
http://svn.digium.com/view/asterisk-addons/trunk/configs/ and it has this
option clearly stated.
Regards
Luis Palma
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On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote:
I don't think so, because in paging/intercom, the phones must support
Auto Answer.
The link you sent says:
SIP phones for the most part don't support any of these phone based
paging functions. If a SIP phone offers an Auto Answer function,
You mean in sip.conf?
Look at adding to your voip providers peer/user config incominglimit,
outgoinglimit or call-limit:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
---
Forrest Beck
www.shift8.biz
On Sep 18, 2007, at 4:26 PM, Jim Boykin wrote:
Is there a way to
On Wed, 2007-09-19 at 01:56 +0530, Jim Boykin wrote:
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
There are
On Wed, Sep 19, 2007 at 01:56:42AM +0530, Jim Boykin wrote:
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
I
On 9/18/07, Atis Lezdins [EMAIL PROTECTED] wrote:
On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote:
I don't think so, because in paging/intercom, the phones must support
Auto Answer.
The link you sent says:
SIP phones for the most part don't support any of these phone based
paging
On Tue, Sep 18, 2007 at 11:27:36AM -0500, Kyle Sexton wrote:
All,
I'm trying to configure queue agents w/ a DUNDi setup so that an agent
can login to whatever server they please w/o any custom setup. In
general this seems to work, agents login w/ AgentCallbackLogin into the
incoming
On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote:
I've stayed out of this thread for a long time, and really didn't read the
past comments, so if I'm repeating someone, I'm sorry. I've been thinking
this for a while, and just have to say it. If you feel like you have to keep
people from
James FitzGibbon wrote:
On 9/18/07, *David Gomillion* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I've stayed out of this thread for a long time, and really didn't
read the past comments, so if I'm repeating someone, I'm sorry.
I've been thinking this for a while, and just
Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon:
On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote:
I've stayed out of this thread for a long time, and really
didn't read the past comments, so if I'm repeating someone,
I'm sorry. I've been thinking
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build sip pvt data for asterisk1/700
Too many open files
Is this a limit of my Linux box? I only
Try:
http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit
On Wed, 19 Sep 2007, Jim Boykin wrote:
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon:
On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote:
I've stayed out of this thread for a long time, and really
didn't read the past comments, so if I'm repeating someone,
Dovid B wrote:
Hi List,
I have a client that has an interesting request. He wants to have
people call in to a conference room and all be able to talk however
they should not hear each other. There should be admin access to for
one user to call in and be able to listen in to everyone that
On Tue, 2007-09-18 at 19:33 -0500, Guillermo Salas M. wrote:
Hi all,
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected
If you have to resort to such measures to get people to work for you
in a motivated fashion, you're doing something very, VERY wrong.
On Tue, 18 Sep 2007, Steve Totaro wrote:
Anselm Martin Hoffmeister wrote:
Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon:
On 9/18/07,
Hi all,
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jim Boykin wrote:
Where do I get sound file for comfort noise. GSM or MP3 is fine.
What kind of comfort noise do you mean? Like background static or music?
If you just want noise (as in pink or white noise), I could make you up
an MP3 or ulaw/alaw
All - I've been wrestling with how to best structure the sip device
accounts on a new asterisk server I'm deploying. All of the sip
devices (currently only Linksys SPA941s) will reside on the same
subnet as the server, and I have already set up a decent automatic
provisioning system for the
Use the extension, and use grep to determine which account uses which
phone. For example I provision my spa9xx phones from a subdirectory on
apache called spa which on slackware is at: /var/www/htdocs/spa/
doing:
grep 123 /var/www/htdocs/spa/* will tell you which phone it is.
On 9/18/07, Erik
Realtime and sip_buddies in mysql works well for very large
installations.
PaulH
On Tue, 2007-09-18 at 22:11 -0500, Erik Anderson wrote:
All - I've been wrestling with how to best structure the sip device
accounts on a new asterisk server I'm deploying. All of the sip
devices (currently
safe_asetrisk bundled with the package, does increase the file limits in
quite a neat way, with some other good setups.
Edit MAXFILES or SYSMAXFILES as required.
Also, I've read posts online, advising not to use safe_asterisk. Any
experiences on this one, anyone?
cheers
- Ben.
Jay R. Ashworth
On 9/18/07, C F [EMAIL PROTECTED] wrote:
Use the extension, and use grep to determine which account uses which
phone. For example I provision my spa9xx phones from a subdirectory on
apache called spa which on slackware is at: /var/www/htdocs/spa/
doing:
grep 123 /var/www/htdocs/spa/* will
Hi Paul,
The way to specify the email_id is as follows
8000 = 8000,ajay,[EMAIL PROTECTED]
Bye and take care.
On 9/17/07, Paul Hales [EMAIL PROTECTED] wrote:
Is there a way to specify multiple email addresses in voicemail.conf for
a specific user?
I seem to remember that it was possible,
The obvious alternative is to use the extension as the sip UID:
Use the extension as the UID and add the mac address as a comment. Like so:
[123]
; Joe Smith
;mac=000E08DA0409
secret = blahblah
... and so on and so forth
This will give the best of both worlds. The mac is readily available and
Thanks Matt, just minimal volume to suite comfort noise.
Jim
On 9/19/07, Matt Riddell [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jim Boykin wrote:
Where do I get sound file for comfort noise. GSM or MP3 is fine.
What kind of comfort noise do you mean? Like
Can someone suggests a good and resonable cost voip provider with
business unlimited plan in USA and allows simultaneous outgoing
calling.
Thanks
~Jim
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