problems. I need your ooh323.conf and all relevant CM config (signal-group,
trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote:
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries
: [asterisk-users] Which H.323 to use in Ast 1.6
Which Avaya system are you running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Wednesday, February 24, 2010 5:52 PM
To: 'Asterisk Users
Subject: Re: [asterisk-users] Which H.323 to use in Ast 1.6
I have always used ooh323 between Avaya and Asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 23, 2010 2:24 PM
To: 'Asterisk Users
We're creating a SIP gateway for a client that will take one leg of a call
in via SIP, and out the other side via H.323. To minimize load on the
gateway, we would like to have the RTP stream bypass the gatewayy altogether
(directrtp/reinvite). Is this possible with these to protocols?
Thanks
We're doing a project that requires H.323 to an Avaya. Does anyone have
experience to share on which H.323 driver to use in asterisk 1.6? Is the
diference between h323 and ooh323 still worth the extra effort? (We've only
installed h323 under 1.4)
If you have setup/config experience with this
: event
17 feb 2010 kl. 23.15 skrev Michelle Dupuis:
Is it possible to just send an event from one Asterisk server to another?
(Perhaps some custom event that I could define?) Or would that break
the SIP protocol/handling in asterisk?
I think this discussion would be easier if you told us what you
To: Asterisk Users List
Subject: Re: [asterisk-users] Access to header field: event
Michelle Dupuis wrote:
I'm trying to pass additional call information (eg: customer ID) to a
call center along with the call itself.
At this point I would be happy just seeing everything that I can get
from
I need to extract the event header info from an incoming SIP call. Is
this accessible from within the dialplan?
I've reviewed RFC 3265 but I'd like to start with just dumping everything to
do with event (if accessible, in other words Asterisk doesn't strip this
away)
Thanks!
MD
--
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, February 17, 2010 4:58 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Access to header field: event
Michelle Dupuis wrote:
*I need to extract the event header info
We have inherited an installation with Ast 1.4 and Aastra phones. The
client complains that sometimes the call audio turns tinny and robotic...I
heard it and it sounds wierd.
Has anyone else experienced this? Cause? Solutions?
Thanks,
MD
--
network) only?
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Michelle Dupuis supp...@ocg.ca wrote:
We have inherited an installation with Ast 1.4 and Aastra phones. The
client complains that sometimes the call audio turns tinny and robotic...I
heard
Please use your quill and ink pot as well, and remember we can't insert
blank paper into the front of a book, only writing on blank pages at the
end.
Oh wait, the advent of computers has allowed us to conveniently insert the
most recent text at the TOP of a message, to prevent people from having
Try unlimitel.ca
-Original Message-
From: asterisk-biz-boun...@lists.digium.com
[mailto:asterisk-biz-boun...@lists.digium.com] On Behalf Of Peter Beckman
Sent: Friday, January 15, 2010 2:13 PM
To: Asterisk Business List
Subject: Re: [asterisk-biz] Looking for 416 DID (Toronto, Canada)
You can address the order of detection problem using udev rules...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Tuesday, January 12, 2010 6:53 PM
To: Asterisk Users List
Subject: Re:
We have that solution running fine...
Is your VPN termination a Linux box? Is it also the office router? Is it
also the firewall?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan McCormack
Sent: Monday,
What do you mean internal timing?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Saturday, January 09, 2010 8:11 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Choppy MOH
-
Could you explain this one a bit more...
You run openSER on the same box as asterisk, and have multiple such boxes,
with the purpose of failover? But if a box goes down with openser on it,
then there is no forwarding. (And most phones can only reg with peer). If
you move openSER to another
I wrote a script to check clients and restart asterisk if registrations died
(external IAX)...but you could modify for your needs. Check it out on
www.generationd.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Are you sure this isn't a Windows zeroconfig problem? If Win drops the
connection while * is talking to your client, the registration could drop
too..
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nic Colledge
Sent: Monday,
List
Subject: Re: [asterisk-users] Can't restart asterisk from script
On Wed, 9 Dec 2009, Michelle Dupuis wrote:
However, I have a cron job that tries to restart asterisk and gets
this
error:
No such command 'restart gracefully' (type 'help restart gracefully'
for other possible commands
I encounter an interesting situation where the internet connection goes down
and then goes back up. The IAX trunks are then unregistered, and * is
confused...only a restart allows * to function again. I have a cron script
that tests for an internet outage and then restarts * after the
obvious - there may also be privilege issues
BillK
On Wed, 2009-12-09 at 22:32 -0500, Michelle Dupuis wrote:
I had double quotes originally - and that didn't work
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
I'm running * 1.4 and can successfully restart asterisk from the command
line with:
/usr/sbin/asterisk -r -x restart gracefully
However, I have a cron job that tries to restart asterisk and gets this
error:
No such command 'restart gracefully' (type 'help restart gracefully' for
other possible
To: Asterisk Users List
Subject: Re: [asterisk-users] Can't restart asterisk from script
Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca
mailto:supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the
command
line
script
Doug Lytle wrote:
Warren Selby wrote:
On Wed, Dec 9, 2009 at 3:08 PM, Michelle Dupuis supp...@ocg.ca
mailto:supp...@ocg.ca mailto:supp...@ocg.ca wrote:
I'm running * 1.4 and can successfully restart asterisk from the
command
line with:
/usr/sbin/asterisk -r -x
variables in the crontab to get it to work.
Billk
On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote:
Interesting...I'll try that. Thanks
__
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
: [asterisk-users] Can't restart asterisk from script
You should replace the single quote with double quote.
--Original Message--
From: Michelle Dupuis
Sender: asterisk-users-boun...@lists.digium.com
To: 'Asterisk Users List'
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
To: Asterisk Users List
Subject: Re: [asterisk-users] Understanding Congestion to incoming caller
2009/11/17 Michelle Dupuis supp...@ocg.ca
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without
We have setup an * box for a small client with 10 phones. They have a
4500/500k ADSL connection which works great when no more than 8 external
calls are in progress. (ulaw)
The problem is when all 10 people try to use an external channel, AND/OR, 8+
incoming calls arrive at once. The symptom
I have an * installation which will refuse incoming callers once a max (5
callers) is reached. Caller 6 and up should be notified of
congestion...without network load on my trunk. How would I do this?
The voipinfo wiki shows playing a congestion tone to the caller, but that
seems stupid since
I'll start with a guess - your asterisk box or firewall is blocking SIP
ports. Diagnose that first (stop iptables/check iptables if unsafe) and try
again...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
You may have to get dirty and run your own multi-tenant box. There's no
clean line between platform which is vanilla PBX without config files and
a multi-tenant PBX with predefined config files (and interface to edit).
You may to be more specific what you expect - otherwise providers might not
That may not work for all sip phones. Some (like xlite/eyebeam) crash when
receiving a text, others drop the subsequent call (Aastra 5x). These
observations are based on a project we did in late 2008; so be sure to do a
proof of concept before you get too deep into the project.
_
From:
Of Alex Balashov
Sent: Monday, November 09, 2009 9:50 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Text messaging
What does Sendtext() actually do? Does it send a SIP request of method
MESSAGE? What does it do on a hardware channel - say, analog or TDM?
Michelle Dupuis wrote
I assume you're kidding?!
RTP is mangled/blocked by most hotspots and mid-size company firewalls...
IAX is often the only way our staff can connect while on the road.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
with this problem
seriously.
I'll take your word for the fact that IAX may be easier, though.
Michelle Dupuis wrote:
I assume you're kidding?!
RTP is mangled/blocked by most hotspots and mid-size company firewalls...
IAX is often the only way our staff can connect while on the road
There is an admin manual you can download from Aastra..have you checked
there? (Not the user manual)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, October 18, 2009 6:18 PM
To: Asterisk Users List
The 57i and 480i are good wireless phones but after 100ft you are out of
range (assuming business interiors). Of you still have to deal with buggy
firmware(and hit and miss tech support).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Thursday, October 08, 2009 10:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] g729 free codec any idea
On 9/10/09 3:31 PM, Michelle Dupuis wrote:
I believe that Intel placed a 729 codec into the public
I like the Qos functionality. Is that a linux based package available for
other distros?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Knight
Sent: Thursday, October 08, 2009 11:15 AM
To: Asterisk Users List
Subject:
And how do you track incoming channels on this trunk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Mathers
Sent: Thursday, October 08, 2009 2:01 PM
To: Asterisk Users List
Subject: Re:
Spinning off from another topic...what are people using for QoS / Shaping?
I'm using Wondershaper script with OK results...but I'd like better. Ideas?
___
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AstriCon 2009 - October
More specificallyI'm looking for a Linux package to allow shaping, QoS,
prioritization by port, etc.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 08, 2009 4:03 PM
To: Asterisk
I believe that Intel placed a 729 codec into the public domain (free), and
someone wrapped it in a nice Asterisk package for use.
No idea where - but I do recall that it is out there, and legal. Of course
it's nice to support a vendor, but free alternatives can't be shunned...
_
From:
In Canada proper grammar requires adding Eh to the end of all questions,
and replacing ou with oo (e.g.: how aboot that hockey game last
night). It's still English, and it's not wrong either.
:)
-Original Message-
From: asterisk-biz-boun...@lists.digium.com
Has anyone written an app that monitors SIP/IAX registration attempts? A
couple of clients are being flooded with SIP registrations (but the source
IP changes every few hours so IPtables won't do)..
I would think that any attempt to reg 5 times with a bad password should
cause a 5 minute
://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html
On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis supp...@ocg.ca wrote:
Has anyone written an app that monitors SIP/IAX registration attempts?
A couple of clients are being flooded with SIP registrations (but the
source IP changes every few
...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, October 02, 2009 2:24 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling
Good post. One of the recommendations is to limit the number of calls per
sip entity. Is there an easy way
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.conf but not iax.conf
Thanks
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 01, 2009 2:27 PM
To: Asterisk Users List
Subject: [asterisk-users] QOS/DSCP for IAX?
Is it possible to set QOS/COS/DSCP on IAX packets? I see some parameters in
sip.conf
-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, October 01, 2009 3:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] QOS/DSCP for IAX?
Michelle Dupuis wrote:
I actually see the TOS setting in iax.conf, but the default (commented
out) is EF - which doesn't
I'm trying to enable res_crypto on a 1.4 installation, but menuconfig says
ssl is needed. I've installed openssl, openssl-devel, openssl-perl
but it's still not happy.
Anyone know what else is needed?
___
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] On Behalf Of Tilghman
Lesher
Sent: Wednesday, September 16, 2009 4:03 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies
On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
I'm trying to enable res_crypto on a 1.4 installation, but menuconfig
says ssl is needed
, September 16, 2009 4:43 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] res-crypto dependencies
On Wednesday 16 September 2009 15:10:57 Michelle Dupuis wrote:
On Wednesday, September 16, 2009 4:03 PM, Tilghman Lesher wrote:
On Wednesday 16 September 2009 14:34:05 Michelle Dupuis wrote:
I'm
I have internal mailboxes that I don't want visible to callers going through
the directory. Is it possible (in * 1.4) to hide mailboxes fom the
directory, without creating a new context?
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Are you using the # symbol in the extension name or to access a feature
(eg: outside line)??
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, September 07, 2009 1:12 PM
To: Asterisk
Start with simple mail testing (forget asterisk)
Does mx1.datagrama.net accept messages for testu...@mydomain.com ? Try a
telnet session first...
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent:
Do a quick search for SMTP commands - to simulate a complete session via
telnet.
Most MTA's will check sender and recipient for validity, relaying, etc. Be
sure both are reasonable and acceptable to host using telnet first.
If you are new to sendmail.cf, read the instructions at the top of
Check your hostname settings, hosts file, and order of name resolution...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bernd
Petrovitsch
Sent: Monday, August 24, 2009 5:26 PM
To: Asterisk Users List
I'm working on a script that needs to determine the extension (eg: 123) of
the phone that initiated the call, or CALLERID number if an externall
caller.
Is there a simple way to do this?
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Does anyone have an example of how to create a custom filename for the
(combined in/out) audio file captured through automon?
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asterisk-users mailing list
To UNSUBSCRIBE or update
Yes - we typically install behind NAT. The issue will usually be your
firewall setup ...assuming you have setup your peers for NAT.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo
Martins
Sent:
Just out of curiosity, how are you planning to use it? (Reading email,
etc?)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of equis software
Sent: Monday, June 08, 2009 7:58 AM
To: Asterisk Users List
Subject: [asterisk-users]
, 2009 2:15 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Suddenly the voice became garbage
(likerobot)using Asterisk 1.4.19.2
Michelle Dupuis escribió:
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun
You're not alone...we never found the cause of this (rare) occurance...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Sunday, May 31, 2009 8:58 PM
To: Asterisk Users List
Subject:
Just check the version of the card (5v vs 3v) - I don't think PCI X is
compatible with the older 5v cards.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, May 27, 2009 9:20 AM
I created a mysql table and lookup script for this. One one server were we
could not use mysql, we created an array of exchanges and compared to those.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean
Pick a release and stick with it as long as you can. Only when you have to
jump, pick a new release, test the hell out of it, and then leave it alone.
Too many people try to keep on the latest release...
_
From: asterisk-users-boun...@lists.digium.com
I'd like to setup a single extension for which all INBOUND and OUTBOUND
calls are recorded to a wav file. I took a look at the wiki:
http://www.voip-info.org/wiki/view/Asterisk+record+calls
but it's not too helpful. Can someone show some sample code in out
recording?
Thanks,
MD
I want to record calls in wav format. Can someone tell me how many MB of
storage per minute each recording requires (assuming SIP / uLaw codec / full
duplex recording)
Thanks,
MD
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I have a simple asterisk install (1.4.18), and want to use call parking. I
can successfully park a call (I see on the CLI that the call is parked to
701). Everything is pretty default.
However, I can't pickup a call from another phone. When I dial 701 from a
phone, asterisk can't find that
, February 25, 2008 8:18 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Parked calls - can't pickup
On Mon, 2008-02-25 at 20:03 -0500, Michelle Dupuis wrote:
However, I can't pickup a call from another phone. When I dial 701
from a phone, asterisk can't find that extensions
Will the built-in T.38 support eliminate the need for spandsp? I'm curious
how this will affect iaxmodem.
MD
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis
Sent: Saturday, February 23, 2008 7:12 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] FXO
Users List
Subject: Re: [asterisk-users] FXO Cards - T38
Michelle Dupuis wrote:
Will the built-in T.38 support eliminate the need for spandsp? I'm
curious how this will affect iaxmodem.
Why on earth would you want to eliminiate spandsp? (which
app_fax from asterisk addons appears
I have setup hylafax today, along with iaxmodem. I'm just starting the
debugging process and see the following message every 60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry:
Restricting registration for peer 'iaxmodem0' to 60 seconds (requested 300)
Can someone
registration for
peer 'iaxmodem0' to 60 seconds
Michelle Dupuis escribió:
I have setup hylafax today, along with iaxmodem. I'm just starting
the debugging process and see the following message every
60 seconds:
[Feb 19 17:44:16] NOTICE[1996]: chan_iax2.c:6021 update_registry
Wow...where did you get that answer?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Paul Hales
Sent: Tuesday, February 19, 2008 8:42 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] ISDN2 facility code...
I have just been given the
Because of our MTA's limitations (Exchange 2007 all-in-one setup), I am
unable to implement a catchall account in Exchange. Does ASSP offer the
ability to rewrite the TO address of an email that the MTA refuses (or ASSP
determines does not exist), sending it instead to a catchall account? (I'm
Just something I noticed: your third line from extensions.conf begins with
s, while the other two begin with _X.
Michelle Dupuis
Technical Support Specialist
Generation Software - Linux and Asterisk solutions and support. Visit us at
www.generationd.com http://www.generationd.com
The Aastra's also have a range of interested firmware bugs that
support/development just can't seem to fix. Do a search for aastra
hang/lockup and you will find what I mean. They look very nice though! For
a simple home deployment, Aastra's are probably great.
-Original Message-
Try access your linux samba box by IP from windows (\\1.2.3.4) instead of by
name (\\servername). Same result?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Behalf Of Greg Sims
Sent: Sunday, January 13, 2008 11:23 AM
To: samba@lists.samba.org
Subject:
I have a AD domain running fine, and I want to add my first linux server.
To keep things simple, I do NOT want to integrate it into my domain. I just
want to create a share that anyone can read/write to.
I've created the setup below which works fine only if I access the linux
share box by ip
Can ALSA send audio to the analog ports AND the optical/toslink port at the
same time? (Or is this a function of the motherboard)?
Thanks,
MD
-Original Message-
From: Clemens Ladisch [mailto:[EMAIL PROTECTED]
Sent: Friday, January 11, 2008 12:49 PM
To: Michelle Dupuis
Cc: ALSA
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
.org] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, January 07, 2008 5:21 AM
To: nut-upsuser@lists.alioth.debian.org
Subject: [Nut-upsuser] Working platform with FreeBSD + USB +
MGE Ellipse(usbhid-ups)
Hi,
___
Nut-upsuser mailing list
Nut-upsuser@lists.alioth.debian.org
http://lists.alioth.debian.org/mailman/listinfo/nut-upsuser
.
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asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz
I've got a Fedora Core 6 system (kernel 2.6.22.14) which automatically
detected my soundcard just fine. The sound chip is and Intel 82801G ICH7
Family (built into motherboard) and has 8 channel audio - which worked great
under windows (so I know wiring is not the issue).
Since switching to
My MB supports both toslink and SPDIF - from what I've read toslink has a
much higher bandwidth. Will the ALSA driver support moving more data
through toslink?
-
Check out the new SourceForge.net Marketplace.
It's the best
Good clue - I changed by kernel options (per the ivtv site) and it now
loads!
Thanks,
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sander Sweers
Sent: Wednesday, January 09, 2008 12:49 PM
To: [EMAIL PROTECTED]; User discussion about IVTV
After a recent upgrade to the latest ivtv (along with kernel upgrade), I
lost my /dev/dvb devices.
What driver should I be looking for (to modprobe) for a Nexus-S ?
Thanks,
MD
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ivtv-users@ivtvdriver.org
I've got a pvr-500mce running fine under Fedora 6 (2.6.20-1.2948.fc6). I
can tune TV channels just fine, but not radio - I only get static.
If I scan I find some channels (see below) which I doubt are real, and when
I tune them I only hear static (see tune result).
I have grabbed the latest
-
From: G. Andrew Walls [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 08, 2008 10:05 PM
To: ivtv-users@ivtvdriver.org
Cc: Michelle Dupuis
Subject: [ivtv-users] Only static using ivtv-radio on PVR-500mce
I've got a pvr-500mce running fine under Fedora 6
(2.6.20-1.2948.fc6).
I can
Well, we can already integrate to major platforms via SMTP. The real value
is in deep integration to the most popular email platform in business:
Exchange.
I would love to see smart Exchange integration, where deleting the VM
attached email will delete the corresponding message from asterisk.
There is a bug in the 480 firmware where if the callerid of the incoming
call is malformed (or basically the Aastra doesn't like, for example have a
# sign in the number), the phone won't ring.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Can someone explain what the facilityenable setting does in zapata.conf
I've read the wiki archive, but it's not even clear what an ISDN
facility is.
Thanks,
MD
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asterisk-users
channel bank?
Exists?
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; DelSp=Yes;
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Quoting Michelle Dupuis [EMAIL PROTECTED]:
We have a client with a Nortel PBX with digital phone
We have a client with a Nortel PBX with digital phone sets. Due to T1
problems (old firmware), we are interested in trying a FXO channel bank.
Is there a channel bank (or equivalent) which emulates Meridian digital
phone sets? In order words, an FXO channel bank that's Meridian digital?
Can someone advise on how to go about finding someone QUALIFIED to make
changes to libpri?
We have a pilot stuck on hold, due to old buggy PRI software on a meridian
PBX. Upgrading the meridian software is not an option, sowe would like
to have libpri changed to compensate for the bug.
Is
read from the cfg file
and set, but it wasn't the case.
Same problem happened to your setup?
On Nov 7, 2007 6:33 PM, Michelle Dupuis [EMAIL PROTECTED] wrote:
Use the web interface of the phone to retrieve the config file that
you uploaded. Is it only partially there?
-Original
Have a look at the smartCID script on www.generationt.com
It allows you to have a database of numbers and override the name (and
number), flag numbers for screening, etc.
MD
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Thursday, November 08,
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