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-
Great job on the new site...
i found some really great people to do some asterisk installs that i needed to
have done for clients through your site hope your new site does well! i'll
be using your site for anything i have in the future for sure.
--
Matt #1
Hello all,
I;m having a (what seems to me) strange problem with some analog lines and
hangup detection.
The site I;m working on has 10 analog lines, my understanding is these are
broken up in 2 invidiaul hunt groups (no idea why, or if this is even true).
I;ve always been told that they
There is a .NET 1.1 library out there... I've played with it a little bit, but
not enough that I could comment on how feature rich or stable it is...
http://www.voip-info.org/wiki/view/Asterisk+.NET
It'll more than likely not be compatible with AMI 1.1 however, which I believe
is included in
Gonzalez [EMAIL
PROTECTED]
Sent: Thursday, April 03, 2008 6:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk
Matt Watson escribió:
There is a .NET 1.1 library out there... I've played with it a little bit
We are using 57i + 560M combination as well... though we are not using the 57i
ct... but the idea of giving them a cordless is a good idea.
The only downside to the Aastra 57i + 560M is that it can only subscribe to 50
extensions for BLF... i haven;t run into this cap yet myself, but I have
Not that I;m complaining But I just got my 2 HPEC license keys from
digium... for TDM800P and TDM400P
asterisk asterisk # zaphpec_enable
Digium High-Performance Echo Canceller Enabler
Copyright (C) 2006, Digium, Inc.
Version 1.0.2
Use the '-l' option to see license information for software
I have a single channel license of Attrafax right now...
It seems to work well from the testing I have done with it so far, which
admittedly isn't as much as I was hoping to have done at this time.
I;m using Linksys SPA2102 ATA's with it... basically what I;m doing is...
FAX Machine -
I'm using Dell 3548P switches currently which I have powering 25 phone (mostly
Aastra 9133i's, a couple 480i's a 57i + 560M)
So... basically I have a phone on about half my ports... my power utilization
on the switch is:
console# show power inline
Unit Power Nominal Power Consumed Power
I can;t imagine what headaches you'd have going from 1.4.11 to 1.4.19.1... that
is a minor version upgrade... no real change in functionality thats
basically 8 versions of bug fixes... if you just apply the IAX2 patch, you'll
be fixing 1 out of probably a hundreds of bugs. Going from 1.4.x
I haven;t used any BRI cards but... call me crazy but wouldn;t they still be
using Zaptel (even your sangoma... the script might just be configuring it for
you)...
and btw, software echo cancel happens in the zaptel kernel driver... it has
nothing to do with the hardware (hence why its a
You might want to begin with tuning your rxgain and txgain settings... there
are a few methods for doing this on the internet, unfortunatly nobody can give
you exactly values to use for tx/rxgain as they will be not only specific to
your install, but specific to every single analog line you
This is my understanding of hyper threading, which I believe to be accurate.
Basically, as some have mentioned previously, the OS 'sees' your single
physical core processor as 2 logical processors, in generally, logical
processors are treated exactly as if they were real processors, and in the
Does anybody know if this version fixes the soft lockup during ztcfg using a
TE200B?
http://bugs.digium.com/print_bug_page.php?bug_id=12468
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development
Team [EMAIL PROTECTED]
err, that should of read TE220B
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Matt Watson [EMAIL
PROTECTED]
Sent: Thursday, May 01, 2008 3:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Zaptel
I don't know about the Cisco phones... I;m using Aastra phones which I can
send a SIP NOTIFY to have them check for updated config... when they detect a
new config they reboot themselves and download the new config.
But your switch might also have an option to disable PoE on a per-port
There is really no reason why you cannot.
Personally... I'd avoid using Java for AGI's that you think are going to
receive heavy use... simply because the JVM adds a lot of overhead, and
possibly a very real performance impact from having the load the JVM everytime.
Of course there is
What Godaddy.com has told you is more or less correct.
Its not their fault that Chinese visitors cannot hit your pages... the internet
is China is highly censored, and quite often they firewall even very large big
name sites like BBC news. Typically they block sites that have any type
discuss
I'm using 1.4.18 in production on 2 boxes... one of which being a custom built
desktop basically, the other being a Dell 1950 III
We are in a migration phase to the Dell box, right now the 1st box is doing
nothing more than being a PSTN gateway to some FXO lines... basically waiting
for
Google is awesome
http://www.voip-info.org/wiki-Asterisk+AGI
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of chetherston miles
Sent: Tuesday, May 06, 2008 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Performance issues
Hello,
I just had to have MTS Allstream fix a new T1 install that we have that we
aren't running in production yet, but it is attached to a production machine.
Apparently they setup the T1 with only a 1 B-channel (how useful!) even though
we had ordered it fully loaded with 23. Anyways...
My bad, I also should of mentioned...
That was on Asterisk 1.4.18 and Zaptel 1.4.10
Using a TE220B
--
Matt
From: Matt Watson
Sent: Tuesday, May 06, 2008 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: PRI D-Channel reconfiguration = crash asterisk?
Hello,
I just
Of Philipp Kempgen
Sent: Tuesday, May 06, 2008 12:27 PM
To: Asterisk Users
Subject: [asterisk-users] This e-mail is confidential ... (was: Re: PRI
D-Channel reconfiguration = crash asterisk?)
Matt Watson schrieb:
Disclaimer Statement: This e-mail is confidential and is intended for the
above-named
It would probably be wiser to run an IMAP server and do imap storage instead of
writing to a cifs-mounted directory... or use ODBC storage... assuming they are
running a database server somewhere.
I don't have any experience with having * write voicemail files to CIFS/SMBFS,
but I also think
There is a enumlookup.agi that is included with FreePBX and thus trixbox, PBX
in a flash, etc. etc.
If you have trouble finding it let me know and I can send you it.
I can;t really vouch for its quality, but I do use it and it does work... but
i;m not sure how well it handles multiple results.
Why are you trying to change the ToS from 46 (0x2e) Expedited for the RTP/RTCP
packets to 16 (0x10)?
I mean... these values really only need to be meaningful to yourself, your
switches, your routers etc however
ToS 46 (0x2e) is the standard value for RTP / RTCP as it is basically the
I don;t have any answers for you...
But I would love to hear about the results after you get this working and what
road blocks you hit and how you overcame them.
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Ex Vito [EMAIL PROTECTED]
Are you using IAX2 as your transport between the 2 servers or SIP?
If you are using IAX2, are you using Asterisk 1.4.18.1 or 1.4.19.1 on either
machine? If so, you may be encountering the IAX2 bug that some have been
discussing on the list recently you can read it here:
I'm a Gentoo guy myself too... but the best advice I can give is just
re-hashing what others have already set... pick whatever you are the most
comfortable with... and if support contracts are important to you, then that
will be a factor as well. I've used most of bigger distros out there over
FreePBX has this functionality... they call it Confirm Calls
I;m not sure if you can set it on actual extensions, but I know you can set it
on ring groups.
I don't imagine the dialplan for doing it is very complicated if you wanted to
do it by hand.
--
Matt
I just took a quick look at the dialplan that freepbx uses for doing call
confirmation... the dialplan part of it is actually quite simple... its just a
matter of setting the USE_CONFIRMATION varialbe =TRUE.
However, the actual magic looks like it happenes through its dialparties.agi...
which
Poking around the zaptel SVN earlier today i see support was added for an
AEX410 card recently...
I'm going to go out on a limb and assume this is the PCI-Express version of the
TDM410?
Any hints on a general availability date?
--
Matt
___
--
I'm not sure if a full-height card would fit (vertically) in a 3U chassis...
but I would probably also assume that if it would not, that the chassis/mobo
would have a PCI/PCI-Express riser card that would mount the cards horizontally.
Might want to check that out with the manufacturer of the
Do you mean
What do I need to configure on my * installation so that only registered sip
users can make calls? ?
If so, you are going to need to give a lot more details regarding your current
configuration for you to get any answers.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL
I'm using Aastra 57i + 560M sidecars for receptionists... the only downside is
that they support a max of 50 BLF subscriptions... you can setup up to 180 blf
keys with 3 560Ms but it will still only subscribe to a max of 50... from what
I understand it's a firmware limitation.
For 4-6 phones
You'd probably want to run something else to handle your registrations like
OpenSER with that many phones.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk Users Mailing List -
Is there any reason you don't want to use Wait()?
However, I would use WaitForRing() myself - its also a great solution on dirty
analog lines where you receive phantom calls.
That being said, I don't know how to do it without using some form of Wait.. as
far as I know zapata.conf doesn't
You can NOT use bogomips as any kind of measurement for system performance.
First of all, Bogomips is a linux-specific thing and not available on other OS
that Asterisk runs on.
The second, and far more important point. Bogo is taken from the word
Bogus. Bogomips are not a measurement of
On May 17, 2008 06:59:43 am Gordon Henderson wrote:
On Sat, 17 May 2008, bilal ghayyad wrote:
Well, why Digium is still using this kind of power
connector while all new machines does not come with
these types?
The new machines that I buy come with legacy power connectors. The flash
IDE
On May 19, 2008 12:51:09 pm Grygoriy Dobrovolskyy wrote:
Hmm, i dont like aastra really much, their transfer management is not human
friendly ;)
What do you mean by that? I've run my Aastra's with BLF using both
Aastra's 'blf' mode and 'blfxfer' mode... the former is basically attended
On May 19, 2008 03:21:34 pm Aaron Stranberg wrote:
Folks,
We are a small office with remote users less than 20 total phone
extensions, and I am looking for some guidance on choosing between
asterisknow and a centos/ubuntu or any other os with an asterisk +
asteriskgui build out? Looking to
On May 19, 2008 06:49:23 pm Kevin Smith wrote:
I almost hate to admit this...but I'm still running Asterisk 1.2 on
Fedora 4 :D
IMO theres nothing wrong with running an old version of * or an old version of
the OS... as long as the box doesn't have a public IP bound to it.
If its an
The first part of this is kind of off topic as it doesn't answer OP's original
question, but instead is a reply to one of the replies.
Cisco is certainly not the only option for doing T38 gatewaying with Asterisk.
I believe Asterisk 1.6 with app_fax supports T.38 origination and termination,
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Machine Options
Matt Watson wrote:
I believe Asterisk 1.6 with app_fax supports T.38 origination and
termination, that is not gatewaying, however if origination and termination
are already there, gatewaying should be fairly
You might want to see if you can change the IRQ assignments in your servers
bios (might have to turn off the PNP OS Installed option if you have one)
--
Matt
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Cavanna, Richard [EMAIL
PROTECTED]
Sent:
Does your extensions.conf have any more configuration than what you've shown?
If not, then you are lacking dialplan for anything but internal calls.
--
Matt
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RoLaNd RoLaNd
Sent: Wednesday, May 21, 2008 9:01 AM
To:
On May 22, 2008 02:06:06 pm Jared Smith wrote:
On Thu, 2008-05-22 at 10:48 -0700, Douglas Garstang wrote:
We didn't want to be generating actual network traffic for this, so I
tried originating a call to [EMAIL PROTECTED]
Why not try [EMAIL PROTECTED] and see if that solves the problem for
On May 22, 2008 04:42:27 pm Steve Totaro wrote:
PS. Figured I would start with DHADI now.
psst. its DAHDI ;)
--
Matt
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
On May 23, 2008 11:25:55 am Dennis P. Clark wrote:
Will fax and dial-up internet work through the gateway?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Carroll
Sent: Friday, May 23, 2008 8:51 AM
To: Asterisk Users Mailing List -
On May 23, 2008 05:27:49 pm Ken D'Ambrosio wrote:
I used to do lots of Asterisk, but got an offer I couldn't refuse, and
went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want
to set up a test system. One thing I'd really like to get my hands on is
recent firmware, etc.,
Asterisk 1.6 probably isn't going to give you any benefit.
--
Matt Watson
http://www.mattgwatson.ca
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: Wednesday, June 04, 2008 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial
In short, fxotune adjusts line impedance, where as adjusting gains I believe
is essentially adjusting the amplification / deamplification of the signal.
http://www.voip-info.org/wiki/view/Asterisk+fxotune
--
Matt Watson
http://www.mattgwatson.ca
On June 6, 2008 12:43:51 am Noah Miller wrote
to attach a fax machine to it...
keep in mind that faxing over VoIP is extremely tricky at best, but if your
entire call path is TDM then you shouldn;t have much of a problem.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation
. I've never tried faxing between ATA's so I
don;t know if they can actually negotiate T.38 support between each other,
but I don't really see a reason why they couldn't.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation
, there is the T.38 protocol which is designed to solve this exact
problem, Asterisk support for it is just rather limited currently (pass-thru
only).
T.38 often gets referred to as FoIP (Fax over Ineternet Protocol)
--
Matt Watson
http://www.mattgwatson.ca
talked about DNS as a possible cause and see if there are any
similarities.
--
Matt Watson
http://www.mattgwatson.ca
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asterisk-users mailing list
To UNSUBSCRIBE or update options
@lists.digium.com
Subject: Re: [asterisk-users] Fax on FXS
On Mon, Jun 09, 2008 at 06:53:34PM -0400, Matt Watson wrote:
On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote:
You should not expect FaxOverVoiceOverIPOverInternet to work well. If
you stick to ulaw codec for the entire call, it might work
don;t need the duplicate group=,
signalling=, switchtype= in zapata.conf
4. you can ditch rxwink= that setting is for non-PRI T1s
try that and see if that helps... I suspect the span not being used as primary
timing source is whats causing your greif.
good luck!
--
Matt Watson
http
they were literally trying to offload their job to this list
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http://www.mattgwatson.ca
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.
All in all it looks like a decent product... i'd be interested in hearing from
anybody that might of been using them for a long period of time (1-2yrs+).
I'm pretty picky about power distribution, i've seen bad power cause too many
problems in my computing history.
--
Matt Watson
http
last post will do load metering...
it'll cost you about twice as much as the one Steve posted however. The sale
price that is, couple hundred more than the regular price.
--
Matt Watson
http://www.mattgwatson.ca
___
-- Bandwidth and Colocation
to its device
classification it could be a web interface like this device, a telnet/ssh
interface, it might not even have any remote capability and just have
physically switches for each port.
--
Matt Watson
http://www.mattgwatson.ca
___
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on it. However I think
that the APC logo on something means alot more than the Sony logo :)
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monitoring server can still page me in the event of an internet outage).
Anyhow, sorry that message got a bit lengthy pretty fast!
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Matt Watson
http://www.mattgwatson.ca
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anything useful.
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Matt Watson
http://www.mattgwatson.ca
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
... if
you compiled from source maybe you just forgot to 'make install' for
asterisk?
--
Matt Watson
http://www.mattgwatson.ca
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
actually have an AstriBank then there is
no sense in even compiling/installing the drivers for it.
I;m guessing you haven't run a make menuselect to select only the drivers you
need?
--
Matt Watson
http://www.mattgwatson.ca
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: Pick whatever distro you are comfortable with
Distro is more of a personal choice than anything... ultimatly they all have
the same software available to them (for the most part), they all just do it
a little bit differently.
--
Matt Watson
http://www.mattgwatson.ca
really just making some logical
assumptions here.
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Matt Watson
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is authentication-less, but its still possible...
and if security is in mind you could limit your TFTP server to specific
source IPs
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Matt Watson
http://www.mattgwatson.ca
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is it only cell phone calls that don't work? or is it any external call
coming in over your lines?
What type of inbound lines do you have? I;m guessing analog lines... if
thats the case what type of signalling are you using?
if its only cell calls and not all external calls then I have no idea
to further develop spandsp.
That being said... i;m also quite pleased to see T.38 support being worked
on for Asterisk... its a pretty important area to further develop IMHO.
--
Matt Watson
http://www.mattgwatson.ca
On Thu, Jul 10, 2008 at 11:54 AM, Steve Totaro
[EMAIL PROTECTED] wrote
libpri if you are just using a TDM400P (since its not a PRI / BRI
[1.6 libpri does BRI as well] card).
Might save you a little bit of time in the future, and its one less thing to
consider as a problem.
--
Matt Watson
http://www.mattgwatson.ca
startup!
Thats on my TE220B. the VPN450 is the hw echo can daughter board.
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I believe HPEC actually is the same algorithm (G.168) that the HW echo
cancel modules use.. the difference being that HPEC uses up CPU cycles and
its performance will be impacted on a system with higher CPU load, whereas
the HW modules have a dedicated DSP for it.
] WARNING[2466]: chan_sip.c:5839 set_destination:
Can't find address for host '72.16.1.20'
[Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination:
Can't find address for host '72.16.1.20'
Might want to post a sip debug of one of the sessions from the Mitel phone.
--
Matt Watson
http
I've seen it before infact there is a website setup where people can
post stuff made with it... kind of super nerdy!
http://www.ratemydialplan.com
--
Matt
http://www.mattgwatson.ca
On 7/27/08, Peter Lindquist [EMAIL PROTECTED] wrote:
Dean Collins wrote:
I just stumbled across this on
I;m using Aastra 480i's 9133i's, 9112i's, and 57i's and none of them have
experienced problems with qualify=yes.
I;m currently on Asterisk 1.4.17, but I've also tested them with 1.4.14 up
to 1.4.19.
--
Matt
http://www.mattgwatson.ca
On Fri, Aug 15, 2008 at 10:59 AM, Drew Gibson [EMAIL
expect very poor results.
3. You are right, you can';t really just make one yourself from scratch, you
need a source that has already been tuned properly to use as a reference for
creating your own.
--
Matt Watson
On Thu, Dec 11, 2008 at 11:01 AM, Olivier [EMAIL PROTECTED] wrote:
Hi,
I
/dahdi.makeopts - but I have not verified
that.
--
Matt Watson
On Thu, Dec 18, 2008 at 11:49 AM, Jerry Geis ge...@pagestation.com wrote:
Jerry Geis schrieb:
/ Is there a way to install DAHDI and NOT download the echo canceler
files?
// I dont have firewall access and its failing.
// I
- it should work just about
everywhere... unless you have total disk encryption or encrypted filesystems
and are unable to mount the partitions... in which case... best of luck to
you.
--
Matt Watson
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Yes, it is available on the SPA2102 - you just login to the web interface,
goto the advanced section, then lan setup... its the very first option.
--
Matt
On Thu, Jan 22, 2009 at 3:29 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
On Thu, Jan 22, 2009 at 3:11 PM, Jeff
On Thu, Jan 22, 2009 at 1:48 PM, Wilton Helm wh...@compuserve.com wrote:
There have been a number of answers provided. The one that was given to
me when I encountered this same problem was to boot a live CD, mount the
root file system and delete the password file which would force your
On Thu, Jan 22, 2009 at 1:52 PM, David @ULC ucoms2...@gmail.com wrote:
I tried :
1. Shut down the machine. (Ctrl+Alt+Del)
2. When it reboot and reach the CentOS boot up screen, then press any key
to go into a select menu. Then press e and navigate to the second line
grub.conf line
On Thu, Jan 22, 2009 at 5:30 PM, Wilton Helm wh...@compuserve.com wrote:
Tripwire would be fine, if it had a baseline, but I don't think its any
good after the fact.
Correct - tripwire does need to be setup beforehand, and its not the most
simple thing to setup *properly*. After the fact...
On Thu, Jan 22, 2009 at 6:05 PM, Wilton Helm wh...@compuserve.com wrote:
making sure to patch any holes through which the hacker might have come
In my case, I had been getting regular attacks through SSH for months,
probably 100 a day (bots). Apparently after nine months of this, someone
On Sat, Jan 17, 2009 at 11:51 AM, Steve Gladden
aster...@michiganbroadband.com wrote:
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP providers that do NOT talk T38 but G711 only.
Does asterisk have the capability to take the T38 call
that way or not... But I haven;t seen anything that would indicate
otherwise.
--
Matt Watson
On 30/01/09 11:11 AM, Jeff LaCoursiere j...@jeff.net wrote:
Following up my own thread, I am kicking myself for quickly posting
without doing a bit of research. Apparently (no surprise) this
integration
Your boss is going to change their mind when they see how awful and
unreliable this setup is going to be lol.
On Fri, Feb 20, 2009 at 1:42 PM, Ignacio sanfermi...@gmail.com wrote:
yep, it is mainly due to cabling issues. My boss doesn't want to
recabling the office.
On Fri, Feb 20, 2009
I find it a little strange that for some reason your box is using includes
located in /usr/local... while there could be reason for this, that seems
like a sign that something might be a little broken on your box.
Also, if you don;t mind me asking...
why would you want to install * directly in
On Mon, Feb 23, 2009 at 9:38 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 24 Feb 2009, David @ULC wrote:
When I am trying to delete voice logs,
[r...@vicidialnow monitor]# rm * -r -f
-bash: /bin/rm: Argument list too long
In the past 30 days, you've asked questions about
The 1.x firmware for Aastra's (for the 9112i / 9133i / 480i) do support some
of the XML functionality that you see in the newer 2.x firmware (for the
more recent models).
I;m not sure if controlling LED status of the keys is supported by 1.x - but
you should be able to find that out by taking a
Not that I;m exactly a big fan of NFS but... why would you choose to
implement a filesystem that was designed to emulate Windows shares for your
UNIX-type environment? You have to kind of expect odd problems like this
when you choose to use things for other than their intended purpose. Samba
I
I'd be interested in this as well... I;m coming up to an upgrade cycle and
trying to decide if I should upgrade to the latest 1.4 or 1.6.1
When others that have commented on this say they have had problems with PSTN
connections, are you referring to T1 or POTS? I;m in a T1 scenerio, so if
There already is a special character to tell asterisk not to parse a line...
its: ; that is why the default configuration is completely filled with
lines that start with ; its considered a comment character to asterisk and
will make it ignore the line... you'd just want to add some extra
I;m not sure how your solution would work... but I thought I'd throw out some
ideas that we are having to implementing faxing here on a new install.
We are going to be bringing in a PRI and routing all the DIDs from our
existing copper lines to the PRI (including fax DIDs)... the solution we
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson
Sent: Monday, December 10, 2007 4:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T.38 fax solution, opinions?
I;m not sure how your solution would work
I'm having this EXACT same problem, I haven;t been able to narrow down the
cause of it yet, but it seems to me that users are receiving notifications
for voicemails in mailboxes that belong to other people, as sometimes their
mail count magically disappears, which I have been suspecting is when
Awesome!
I was an Attrafax customer and was very disappointed when it vanished and
couldn;t get new modules for newer versions Asterisk with our paid license.
If anybody is working on t38 gatewaying code for 1.6, it would be worth a
look at this, as I can attest that Attrafax worked quite well
I just downloaded a copy of this, by any chances does Zoiper by any chance
have diff files available for a more recent 1.4.x release? (I know 1.6 is
probably out of the question)
Thanks,
--
Matt
On Mon, Mar 8, 2010 at 12:11 PM, Matt Watson m...@mattgwatson.ca wrote:
Awesome!
I
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