To solve the problem of Local channels answering Queue calls, I thought
about myabe using a channel variable switch that turns on before the Queue
is called, and a check in the extension-dial to Local/ext to see if it is a
queue call and shouldn't go to voicemail, or if it's just a direct call
Heres a little teaser for those of you with iPhones
Asterisk Voicemail for iPhone allows you to check your voicemail
messages on your house or business line from your iPhone. You can
think of it as Visual Voicemail, but for your Asterisk PBX numbers
instead of your ATT cell number. The technology
across the issue (and it's an issue
with this one because the client expects the external caller ID).
Are there any ways around it?
Thanks in advance.
Regards,
Chris
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Royce Souther wrote:
Are there any tests that can be done to pinpoint the problem?
Swap out the card - that usually fixes anything you have control over.
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other GUIs (free or commercial) that reliably
support attended transfers?
I'm trying to replicate as much of the Avaya Phone Manager application in
asterisk as possible, and really finding it quite a struggle.
Thanks in advance.
Regards,
Chris
bilal ghayyad wrote:
OK that worked, but how can we resolve it without need
to type the command manually, as the destination might
change its IP address without our notice, so the
question is:
How can the host be updated periodically (like
externrefresh settings), but need it for host, any
Proxy - sip.essex1.com (10.1.3.2)
Isn't it a bit unusual for their proxy to be given to you as an RFC1918
address? Unless you're on their LAN of course...
Regards,
Chris
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desk).
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Chris
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If there's relatively little cost difference between the two (factoring in
necessary PC redundancy requirements, etc.) then it really comes down to which
do you feel more comfortable configuring?
Regards,
Chris
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?
Regards,
Chris
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with a number of S450s to use a common phonebook and keep them in
sync.
Regards,
Chris
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I just get one of the older models?
From what I've been told it's probably still a few weeks away, so if you have
an urgent requirement, I'd go with the S450 for now.
Regards,
Chris
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should be able to port your number(s) on
your analogue lines over to your IP trunk provider.
Regards,
Chris
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the thing regularly, you'll
be grateful for the improvement.
I'd like to get my hands on a Snom M3 to test, but over here in the UK it's
nearly 3x the price of the Siemens S450,so I fear customer uptake will be
limited at best.
Regards,
Chris
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For full
unreasonable load on your * server, try dropping the registration interval down
to something small like 300 (5 minutes), and disable STUN entirely (obviously
making sure nat=yes is defined in sip.conf for those devices).
Regards,
Chris
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For full
) with ztdummy. We've had
plenty of meetme conferences over that time and I've not noticed any problems.
I'll run zttest on a couple of them over the weekend and see what results we
get, but certainly, there haven't been any complaints about quality during
conferences.
Regards,
Chris
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under admin/advanced/Phone menu on
the 942. It does not seem to pick it up automatically from asterisk like many
phones do.
If you’re configuring a lot of similar handsets, consider using an
autoprovisioning script - it'll save you a hell of a lot of time in the long
run.
Regards,
Chris
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) will *not* result in
encryption when encryption=yes is set?
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Chris
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If you've longed for a text-to-speech Asterisk toolkit that sounds
just like the default Allison prompts that ship with Asterisk 1.4,
then today is your lucky day.
Havn't tried it myself but hopefully it comes close to what you need
it for.
Regards,
Chris Bennett
,
Chris
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proximity detection. Never configured it myself but I've
always liked the idea of it :)
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Chris Bennett
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settings can be
managed from our web interfaces should configs change in the future.
I'll log into the tftp server and grab the config files in question if they're
likely to be of any use and post sanitized versions of them to the list later
tonight.
Regards,
Chris
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VoicemailMain(${CALLERID(number)}) is probably what you want.
Regards,
Chris
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it'll never be able to access /dev/zap/*
FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth
updating to 2007.0 if you can spare the time - it'll save you a lot of messing
around with gcc versions etc. later down the line.
Regards,
Chris
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many hours playing with echo
cancellation and gain settings on the FXS module.
Regards,
Chris
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-standard figures through a provisioning system?
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Chris
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it that way
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fixed drasically in asterisk 1.2/1.4
anyone know?
Ideas/suggestions appreciated ...
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Thomas Kenyon [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Is there an advantage to having a Queue members URI in the form:
SIP/User (or indeed IAX2/User)
Over
Local/number@context
mark - any thoughts on this?
If anyone knows a good Polycom wholesaler in the UK, I'd be really grateful if
you could fire me some contact details off-list, or post them to the -biz list
(to avoid cluttering this non-biz list).
Thanks in advance.
Regards,
Chris
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I'll third the Polycom units. We use SoundStation IP 4000s in our
conference rooms and they work great. But yes, they are expensive.
- Chris
On 24 Dec, 2007, at 7:20 AM, Michael Graves wrote:
On Mon, 24 Dec 2007 14:29:46 -, Chris Bagnall wrote:
Greetings list,
Does anyone have
Is there anyway to code in the Asterisk dialplan to show BOTH lines are busy
when either of 200 or 201 are in use?
exten = 200,hint,SIP/200SIP/201
exten = 201,hint,SIP/200SIP/201
Regards,
Chris
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asterisk to use an ephemeral source port for its outbound IAX2 connections...
Chris
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should do what you're after.
Chris
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the softphone reg info
between sites? Or is there some DUNDi magic that would allow a softphone
from site-A to register on site-B's asterisk box without site-B having a
local entry in iax.conf for that softphone?
Chris
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? Is there an obvious solution here
that's escaped me?
(Ugh, why couldn't iaxclient/zoiper/asterisk all just follow the
RFCs and use ephemeral source ports to begin with?)
Thanks,
Chris
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portage tree.
This is something I'd consider paying an annual subscription for.
Regards,
Chris
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didn't consider call parking... I'll see if that is suitable for the
users..
Anyone else know a better way to do this?
Would be good to hear any other ideas.
Thanks,
Chris Bennett
(cgb)
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debugging I can do to determine the cause of
voice quality degradation? Is there a way I can configure the
asterisk servers to not communicate the RTP traffic across the DSL
links and back again?
Any suggestions will be much appreciated.
Regards,
Chris Bennett
as
configured, but it isn't, and ive no idea why.
Hope you can help
Chris
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Ive got a linksys SPA-3102 on order to replace this card...
but i would still like to confirm if it is the card thats duff or
something with my config, more of a geeky pride excercise than
anything else.
Cheers
Chris
On 11/12/2007, Drew Gibson [EMAIL PROTECTED] wrote:
Chris Boczko wrote:
Hello
kill any numbers starting 07 with a
quick string comparison function.
Regards,
Chris
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connection simultaneously, but only if it's not
going to cause problems with the rest of the setup.
Any suggestions gratefully appreciated.
Regards,
Chris
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I have a multi-homed setup, and haven't had any issues, though it's
two separate network segments.
What version of * are you using?
Regards,
Chris
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You can also take a look at the T.38 product from Attractel.
http://attractel.com/faxterisk.php
Disclaimer, I work for these guys.
Chris
On Sun, November 25, 2007 4:11 pm, Robert Moskowitz wrote:
Olivier wrote:
Robert,
Do you mean T.38 passthrough ou T.38 to T.30 gateway ?
The former
);
Then in the dial play did something like Set(CALLERID(number)=${myvariable})
It may not be the most elegant solution but it works fine for me.
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: 18 November 2007 16:48
To: Asterisk
and make sure a sensible
response comes back).
Regards,
Chris
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or work around this error.
Chris
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Jarga Jallow wrote:
I need help with my grand stream GXP2000 phones they keep freezing
randomly. Any ideas?
What firmware revision?
Want to buy a used one from me? I'm trying to standardize on Sipura 841s,
and I have
one GXP2000.
Jarga
--
Chris 'Xenon' Hanson | Xenon @ 3D
I started out a few years ago with some SPA-841 sets, because the
Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more
call appearances, and I didn't want just the 4 max that the SPA offered. As
it turns out, with the greater flexibility of VOIP, I don't need 'dedicated'
Hi all. Newbie to the list, been using VOIP with Sipura Grandstream
hardphones for a few years, via a VOIP service provider (who I won't name
here). I haven't stepped up to running my own Asterisk box yet, because of
poor reliability of our Internet connection during non-business hours, but
I'm
, etc.) and we haven't had any problems with them. Would be interesting to
know how well it'll scale with more VMs on each box.
Regards,
Chris
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2-3 concurrent calls, so I'd be very surprised if
it's a load issue.
If it makes any difference, asterisk is 1.2.17 (same version as on most of our
other boxes), hardware is a Via C3 (Nehemiah) with 256MB RAM.
Any suggestions gratefully appreciated.
Regards,
Chris
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Use a linksys speakerphone behind a metal late and a Push to Connect
button wired as the hook switch - bat phone connection. Mount the mic
and speaker on holes in the plate and the guts glued to the plate.
Simple and cheap and they have to buy from you.
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a discussion about other
projects on any of those mailing lists.
So, are there any mailing lists out there offering a non-project-specific
discussion of the options out there and when use of each is most appropriate?
Regards,
Chris
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:-)
Regards,
Chris
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asterisk
If I upgrade libpri 1.4.0 to 1.4.1, do I then need to recompile
asterisk even though I'm not upgrading asterisk?
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615-221-4200 x103
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Chris Bagnall wrote:
The 4 line limitation has never been a problem for the vast majority of
people.
I can't imagine what an office worker would do with four line
appearances. I use a 6 line Polycom but I register different line
appearances to different customer PBX's that I am working
What version of ODBC does asterisk 1.4 need?
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: *** The unixODBC installation on this system appears to be broken.
configure: *** Either correct the installation, or run configure
configure: *** without explicitly specifying --with-odbc
--
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[EMAIL PROTECTED
The libtool-ltdl package is installed.
On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote:
I'm having an error when I try to ./configure asterisk using
--with-odbc=/usr/lib. Below is the version of each application and the
./configure
.
On 10/1/07, Chris Stinson [EMAIL PROTECTED] wrote:
The libtool-ltdl package is installed.
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to be an ATA with an FXO port (e.g. Sipura/Linksys
SPA-3000/3102), the other device can be either an ATA or a SIP Phone.
Does anyone have any hardware recommendations that'll work in this scenario?
Regards,
Chris
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finally get rid of them all.
[if anyone in the UK wants some second-hand GXP2000s I have quite a few, about
18 months old, in good condition :-) ]
Regards,
Chris
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involved.
Regards,
Chris
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fairly well.
I've replaced the Cisco 7960 on my desk with the colour SPA-962. It's about
half the buy price of the Cisco, takes less desk space, has more features, and
a vastly superior screen.
Regards,
Chris
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of people.
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Chris
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for this type of service. Why don't you take a pay as you go
plan and pay for what you use?
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It might be simpler to have the person record the message then attach it
to an extension. The audience calls the number/extension and listens to
the broadcast.
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Yahoo
Kenneth T. Van Wie II wrote:
...
Did you mean to include an answer or a question of some type?
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the
features you did not want and to enable auto-answer out of the box for a
fee.
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[Sep 16 06:20:44] NOTICE[18423]: pbx_spool.c:371 attempt_thread: Call
completed to Local/[EMAIL PROTECTED]
This setup worked without problems under asterisk 1.2.
Any idea of what's going on? I'd be happy to provide more
information if needed.
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Email: [EMAIL PROTECTED]
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that be the issue?
I tested with an AGI script, and the problem is that audio isn't sent.
The script receives DTMF digits and otherwise acts as expected.
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On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
On Fri, 14 Sep 2007 20:46:13 -0400, John Albano [EMAIL PROTECTED] wrote:
I'm seeing the problem on both etch and lenny releases.
Linux ads04 2.6.18 #2
On Sun, 16 Sep 2007 01:29:11 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 01:30:26PM +, Chris Nestrud wrote:
On Sat, 15 Sep 2007 16:03:07 +0300, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sat, Sep 15, 2007 at 12:47:59PM +, Chris Nestrud wrote:
On Fri, 14 Sep
of what is causing this problem and how it can be solved?
Chris
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still, put the phones on the same network as the Asterisk PBX and
say goodbye to your problems.
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with the Alcatel unit? Any obvious pitfalls to
watch out for? Any suggestions gratefully appreciated.
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Chris
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even if it isn't open source, there isn't any $ expense)
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Chris
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here and I've yet to find
any worthwhile feature differences.
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Chris
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in the pre-compile
configuration?
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oohs no!
Whats up, haven't heard much out of you lately.
Chris
Brian West wrote:
Everyone,
I will be attending Astricon in Phoenix and would like to have a
little get together to discuss Open Source Telephony and the
challenges we as developers and system integrators
that was the issue.
Steve,
How are you providing surge protection? I have lost a couple of cards to
storms also.
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anyone suggest a way round this?
Thanks in advance.
Regards,
Chris
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Digium has a handy tool online!
http://www.digium.com/en/products/voice/audioconverter.php
:-)
--
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Alex Balashov [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On Sun, 12 Aug 2007, MOSBAH ABDELKADER wrote:
Hello all,
have anyone an idea about converting an audio file
command to grab any ZAP or IAX channel
--
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Carlos Chavez [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
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Watch the mpg123 process, it can take 99% of available cycles and slow
everything down.
I would disable http, cups, smb and any other non-vital process, reboot
and see if things are better.
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.
SIP Phone -- Asterisk-- Linksys SPA3102 -- PSTN (In Use) --
Use IAX
Can any one help me with some dial plan logic for this; I'm confused as to
the best way around this?
Thanks in advance
Chris
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Great idea/strategy, thank you!
Going to be c00l if I can get this working
regards,
--
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Jared Smith [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
On Thu, 2007-07-19 at 09:16 -0400, Chris Earle wrote:
Trying to do what should be a basic info retrieval from my asterisk
You can also give the our T.38 stack a try.
http://www.attractel.com/t38.html
Chris Childress
AsteriskGuru.com
Andrew Joakimsen wrote:
I have already tried to contact to persons from Digium and I did not
receive a response.
I was wondering if there is any plan to support fully faxing
! ;-)
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Senior Developer
We are a fast growing service provider to the small and medium sized
business community in NYC. The company is looking for a senior developer to
lead the next stage of development for a web-based service application. The
opportunity is significant for the right
Is there any way to set the targeting ip that is sent out in the
dundi answer (to my public ip or any other where i want to receive the
call)?
Change your mapping in dundi.conf to reflect your true public IP rather than
using ${IPADDRESS}.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur
Lee Jenkins wrote:
I'd say that Micro is the MS of Restaurant POS. We replace their
systems regularly ;)
I'm curious what with?
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305) 704-7249 Fax: (815)301-9759
Yahoo IM only: [EMAIL PROTECTED
phone before they leave the office? Depending on the device,
they can either do that locally on the device, or alternatively, you can
program a couple of short codes into your dialplan to allow the client to
enable/disable divert.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
Teliax has been great, VoipJet seems to be working vry hard to give
outstanding reliability and fault notifications so I would recommend
them as a backup. I like to have two account and failover for termination.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
and Canadian
border.
--
Chris Mason
Anguilla: (264) 497-5670 Fax: (264) 497-8463 Cell: 264-235-5670
International: (305) 704-7249 Fax: (815)301-9759
Yahoo IM only: [EMAIL PROTECTED]
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I love the smell of lemonade in the morning
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Chris Mason
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Jason Ma wrote:
Hi guys,
Does anybody try to install IPV6 support on asterisk?I just found a
patch for that but it is released on 2005,I have no idea if there is new
version to support ipv6 or new patches,please advise.Thanks a lot.
It is a very desirable feature that will solve a lot of
you may be forced to
use g729 simply because so few hardware devices support speex.
It'd be interesting to see some comparisons or comments from people using g726
as this does seem to be supported by quite a few hardware devices.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
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