Sorry about the trouble. Unsubscribed that user from the mailing lists.
Matthew Fredrickson
On Fri, Aug 7, 2020 at 9:20 PM Elizabeth wrote:
>
> I'm online on this site!
> So contact me in my profil
On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote:
>>
>> There is a big confusion here about Stir Shaken. It is NOT a provider issue.
>> Un fact, all providers are whasing their hands and modifying their swihtches
>> to pass-through the Signature. They cannot sign the call because then the
ocused more on the Asterisk 18 side of things
rather than clarifying a lot of that :-)
Matthew Fredrickson
> --
> Joshua C. Colp
> Asterisk Technical Lead
> Sangoma Technologies
> Check us out at www.sangoma.com and www.asterisk.org
> --
> __
Hey All,
For those of you that do not know me, my name is Matthew Fredrickson
and I’m the project lead for the Asterisk project. First off, I wanted
to thank all of you that contribute in various ways to the project –
whether it be at a developmental level, answering questions on forums
his
> doable? If not possible with PJSIP, is chan_sip a better option? Any
> pointer would be greatly appreciated.
>
Right now, chan_pjsip does not properly handle tel: URIs. If you need them
you might need to use chan_sip.
Matthew Fredrickson
>
> Thanks,
> --
> Jean-Denis Girard
>
Unsubscribe info is in the footer of the message
Best wishes,
Matthew Fredrickson
On Mon, Oct 1, 2018, 6:22 AM Karen York wrote:
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
eb page but I never
> could get Cyber Mega Phone 2K to work on the same server. We used JSSIP
> to create the webrtc phone on our website.
We just updated the documentation for how to get CMP2K working on the
wiki [1]. We'd love some feedback if you still have issues getting it
setup s
Hey,
I would suggest starting a new thread with this question instead of
inserting this into another existing thread like this.
Matthew Fredrickson
On Tue, Sep 18, 2018, 11:16 AM modou lo wrote:
> Please can i ask you i want to know which code can help me to provide the
> taxation o
Usually yes. You'll need to read the UPGRADE.txt and CHANGES files to get a
good idea of the specific changes though.
Best wishes,
Matthew Fredrickson
On Thu, Sep 6, 2018, 7:44 PM Telium Support Group wrote:
> Does anyone know if Asterisk 16 includes changes to the AMI? (syntax /
> co
May 22, 2018 at 7:58 PM, Matt Fredrickson <cres...@digium.com>
> wrote:
>
>> More testing. Test test test. :-)
>>
>> --
>> Matthew Fredrickson
>> Digium, Inc. | Engineer
Here we go!
Matt
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
/security/IAX2-security.pdf
With regards to the CTOKEN addition. Hope that helps.
Matthew Fredrickson
Digium, Inc.
On 3/8/13 8:38 AM, Thorsten Göllner wrote:
Hi,
I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine.
But 1 thing will not work: IAX. I used the same
try removing that and retry your outbound call test.
Hope that helps a bit.
Matthew Fredrickson
Digium, Inc.
On 2/11/13 10:54 AM, Kevin Wright wrote:
I forgot to add, cat /proc/dahdi/* yields:
Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER)
1 WCTDM/4/0 FXSKS (In use) (EC: MG2
=11,3,0,esf,b8zs
span=12,4,0,esf,b8zs
Hope that helps.
Matthew Fredrickson
Digium, Inc.
On 12/20/12 10:42 PM, Dave George wrote:
I have a box with 12 T1s (4 Te410P cards). The PSTN provider is
reporting slips and ask me to update the clock source. I have my
system.conf set as the following
- at least not yet)
Thanks, Vieri --
That's the reason why it's not working. Unfortunately, the newer
versions of those cards require DAHDI in order to operate.
Matthew Fredrickson
Software/Hardware Engineer
Digium, Inc
in it, for reference on configuration.
Matthew Fredrickson
Hardware/Software Engineer
Digium, Inc.
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On 11/18/10 10:07 AM, Matthew Fredrickson wrote:
On 11/18/10 7:40 AM, Matt wrote:
On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com
wrote:
On 11/17/10 2:44 PM, Cary Fitch wrote:
In regard to #2, any T1 card should work. But the problem is you need
SS7
software
obviously have used it with
quite a few Digium cards that have worked well.
Matthew Fredrickson
Hardware/Software Engineer
Digium, Inc.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt
Sent
the message. This is just libpri re-establishing layer when
the other side tries to drop it, due to its desire to have the
perception of a persistent layer 2 (in older versions).
In newer libpri (1.4 branch) it allows layer 2 to drop and stay dropped
until it is needed by layer 3.
Matthew
not sure what my schedule is going
to be like there, but I'd love to hear about any meetups that may happen
if I can fit it in.
Matthew Fredrickson
Digium, Inc.
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Martin wrote:
pri debug span 1
should show you the ISDN messages for 2BCT if there are any
Also someone should have told you that the 2BCT code is by default not
compiling
and you could enable it by editing chan_dahdi.c and adding
#define PRI_2BCT
Also since this flag is not
for it to work, 'transfer=yes' must be set in chan_dahdi.conf
on each of the channels you would like to enable it on.
To disable it for a channel or group of channels, set 'transfer=no'
above that group.
Matthew Fredrickson
Digium, Inc.
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did not add support for
this feature since it would probably have required some core changes to
do so. So right now, we simply ignore that message and go about our
merry way.
Matthew Fredrickson
Digium, Inc.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y
what the problem is.
Matthew Fredrickson
Digium, Inc.
-- Native bridging DAHDI/1-1 and DAHDI/3-1
Protocol Discriminator: Q.931 (8) len=28
Call Ref: len= 2 (reference 801/0x321) (Terminator)
Message type: FACILITY (98)
[1c 15 91 a1 12 02 01 23 06 07 2a 86 48 ce 15 00
it) :-)
--
Matthew Fredrickson
Digium, Inc.
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or something of that nature.
The most current version of fxotune is pretty much immune to dialtone or
other background noise due to the newer way it does signal measurement
(using frequency analysis instead of frequency agnostic power
calculation), so you shouldn't see any problems with this.
Matthew
Zaptel though,
you can build use the fxotune utility from the latest version of Zaptel
and just don't run make install so you don't overwrite your existing Zaptel.
Matthew Fredrickson
Digium, Inc.
Any advise.
Regards
Bilal
--- On Thu, 4/2/09, Matthew Fredrickson cres...@digium.com
signaling and bearer channels on the
same box.
Matthew Fredrickson (the libss7 guy :-) )
Digium, Inc.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of lizhong zhu
Sent: Friday, March 20, 2009 2:05
anyone make asterisk work in TE/PtmP with a B410P ?
Hey Olivier,
I wrote most of this code, and would be very interested in asking you a
little more about this issue.
Can you IM me?
On MSN, I am creslin...@hotmail.com
AOL: MatthewFredricks
jabber: cres...@digium.com
Thanks,
Matthew Fredrickson
Olivier wrote:
2009/2/27 Matthew Fredrickson cres...@digium.com
mailto:cres...@digium.com
I have a couple of suggestions:
Make sure that your timing configuration is correct in
/etc/dahdi/system.conf (that it has a valid timing source).
Also, you probably
, that would be a good thing.
Matthew Fredrickson
Digium, Inc.
fax nsf 00
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through
g711alaw
no vad
And upon further examination... don't put T38CALL in as a variable. It
will cause the initial INVITE to only
have T38. Leave
(which disables the EC) takes a really
long period of time to happen, and if it does, disabling the EC in the
middle of the fax will usually cause a fax failure.
Matthew Fredrickson
Digium, Inc.
Olivier wrote:
2009/2/25 stoffell stoff...@gmail.com mailto:stoff...@gmail.com
Hi all
your question.
The likely answer is that we probably do not decode/expose this
parameter to the dialplan at this time, but adding and exposing
parameters is not a very hard thing to do.
Matthew Fredrickson
Digium, Inc.
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that NT-PTMP support
might be added, but I could not give you a concrete time at which it
will be done, since it will probably require some significant internal
changes in libpri.
To answer your final question, for now, if you need NT-PTMP mode, you
should use mISDN.
Matthew Fredrickson
Digium
Patrick wrote:
Matthew Fredrickson wrote:
[snip]
I actually was the one that did a lot the work in adding the BRI support
to libpri/chan_dahdi.
[snip]
To answer your final question, for now, if you need NT-PTMP mode, you
should use mISDN.
Hi Matthew,
Is there a BRI status document
Bob Pierce wrote:
Has there been any work done on using Asterisk as an MGCP client?
Nope :-( Still a no go.
Matthew Fredrickson
Digium, Inc.
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Matt Riddell wrote:
On 18/11/2008 9:46 a.m., Matthew Fredrickson wrote:
Singer X.J. Wang wrote:
We've had the same issue. For calls that go between a SIP connection
(desktop phones) and Zaptel connections, there was a lot of problems
with half duplex. We switched
from the Digium card
a Digium
card, but were in fact sold an OpenVox card which did not perform as
well as would be expected from our cards, causing some grief and
confusion for them. It can be confusing because they use the Digium
driver and look like a Digium card to the driver.
---
Matthew Fredrickson
Digium
processor caches.
Matthew Fredrickson
Digium, Inc.
Wilton
Wilton--
AFAIK, the current algorithms (old new) are indeed table lookup.
It wouldn't hurt for you to do a code review on them, you might
be able to improve them...!
murf
For those of you interested in a slightly longer
boards, Sangoma
uses the exact same hardware echo canceller.
Matthew Fredrickson
Digium, Inc.
Doug Lytle wrote:
Ken Williams wrote:
We’ve had an issue since we went live nearly two years ago on Asterisk
where people complain about not being able to talk while someone else
is talking. I
Benny Amorsen wrote:
Matthew Fredrickson [EMAIL PROTECTED] writes:
Actually, with the way caching is done on nearly all modern processors,
it is debatable whether or not a look up table is the optimal way to do
the conversion, at least on such a simple codec such as ulaw or alaw.
In fact
Steve Underwood wrote:
Matthew Fredrickson wrote:
Actually, with the way caching is done on nearly all modern processors,
it is debatable whether or not a look up table is the optimal way to do
the conversion, at least on such a simple codec such as ulaw or alaw.
In fact, the amount
Steve Underwood wrote:
Matthew Fredrickson wrote:
Actually, with the way caching is done on nearly all modern processors,
it is debatable whether or not a look up table is the optimal way to do
the conversion, at least on such a simple codec such as ulaw or alaw.
In fact, the amount
processed
---
Matthew Fredrickson
Digium, Inc.
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Steve Totaro wrote:
On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Sebastian Gutierrez wrote:
Anyone is using 1.6 in production??
Is it ready?
I have a number of people using 1.6 in production doing
original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew
Fredrickson
Enviado el: Friday, November 07, 2008 3:18 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] 1.6 Production ready??
Steve Totaro wrote:
On Fri, Nov 7
original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Matthew
Fredrickson
Enviado el: Friday, November 07, 2008 3:18 PM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] 1.6 Production ready??
Steve Totaro wrote:
On Fri, Nov 7
in that
SETUP message below. :-)
Matthew Fredrickson
Digium, Inc.
Please help me out.
My zapata.conf
[trunkgroups]
[channels]
context=pstnincoming
pridialplan=local
it into it, but, any time you rewrite code or do something new,
there's always going to be a period of shaking out of unforeseen bugs.
Sorry if you have had any trouble. The name change and related efforts
have been just as hard on us as developers as it has been on people that
use it.
--
Matthew
; where pri_ is either pri_net or pri_cpe
channel=1-23 ; or whatever your channels are associated with the PRI.
Matthew Fredrickson
Digium, Inc.
eg.
This works:
[from-pstn]
exten = _34397333,1,Dial(SIP/511,30)
exten = _34397333,n,Hangup
This also works:
exten = _34397333,1,Dial(SIP
Jay R. Ashworth wrote:
On Mon, Sep 08, 2008 at 11:28:13AM -0500, Matthew Fredrickson wrote:
For DMS100's version of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Ok
Jay R. Ashworth wrote:
On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote:
Will I actually need to do PRI debug on that span to tell?
Or will seeing hangup messages while I'm still talking be the solution?
Seeing hangup messages on the console while the audio path remains
Jay R. Ashworth wrote:
On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote:
Let me clarify some of this.
Under no circumstances can Asterisk receive a TBCT request. We just
ignore them. We can initiate them however.
There are different TBCT implementations, dependent
successful and high traffic installations around the world.
The current record (that I have been told of) is an installation doing
over 100,000 calls per day. So try to beat that ;-)
Matthew Fredrickson
Digium, Inc.
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of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Hope that helps a bit.
Matthew Fredrickson
Digium, Inc.
___
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emist wrote:
My best guess from looking at that is that its a driver bug. The last
thing that happens before the lockup seems to be an ioctl call to the
device.
That was a bug that should have been resolved by 1.4.11 (he subsequently
updated and it was resolved).
Matthew Fredrickson
should be fine with a TE412. Just make sure that your line is
plugged in correctly and your span= line is correct for the line settings.
Matthew Fredrickson
Digium, Inc.
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AstriCon
a DISCONNECT is received
with Inband progress information avaiable.
Matthew Fredrickson
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk
using the second set of
settings you listed above. Which echo canceller are you using with
this, by the way? (Hardware, software, if software, which software echo
canceller).
Matthew Fredrickson
John
On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
John
is pre-hardware echo canceller and txgain is post hardware echo
canceller. (zapata.conf rxgain and txgain).
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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the attachment didn't go through. Can
you email the tarball to me directly or post it to a website?
Thanks,
Matthew Fredrickson
line 7 during this run. fxotune.conf now contains:
5=7,255,251,251,2,255,255,1,255
6=7,255,251,251,2,255,255,1,255
7=4,0,0,0,0,0,0,0,0
8
with the calibration process.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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).
Matthew Fredrickson
John
On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 04 June 2008 22:02:19 John Morey wrote:
Hello,
I've run fxotune at different times but continue to get what seem to be
strange numbers in /etc/fxotune.conf. It ends up
.* specs to get it working, that
will be a bit more trouble.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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many channels on it.
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Software/Firmware Engineer
Digium, Inc.
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installed on the
system that's used as the test machine.
This was a regression do to recent Makefile changes. A test for this
problem has now been added to our pre-release regression testing.
Matthew Fredrickson
sigh.
Thanks,
Steve Totaro
On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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to this type of
problem, but probably the best solution is to call support and talk to
them about this.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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asterisk-users
check hardware compatibility
before you buy anything.
For the purposes of making sure list records are accurate, this in not
true. Asterisk was indeed written with the intention to run on
multi-core systems, and should utilize extra cores just fine.
--
Matthew Fredrickson
Software/Firmware
Ex Vito wrote:
On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson
[EMAIL PROTECTED] wrote:
Ex Vito wrote:
Matthew,
...is there any specific test you'd like us to perform on this revision
?
(considering that currently we have no PSTN line to attach to... we
can
Ex Vito wrote:
On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
Ex Vito wrote:
Tested with no 4K stack kernel and stackcleanup svn branch
zaptel version. Correct, the kernel no longer complains about
the soft hangup.
However the system still hangs (console
Ex Vito wrote:
On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
I just realized where this is coming from. I was attempting to patch
this from a different angle, but as soon as you mentioned the drastic
difference in load time I realized what had happened
Ex Vito wrote:
On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote:
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
I just updated the branch. Wait about 5-10 minutes in case for the
changes to get mirrored, and then try updating and doing
of the
quad span cards do not advertise support for.
2008-03-14 16:39 + [r3983-3990] Matthew Fredrickson [EMAIL PROTECTED]
* firmware/Makefile, kernel/wctdm24xxp/base.c,
kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update
wctdm24xxp's VPMADT032
Ex Vito wrote:
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
The softlockup indicator should be benign. It gets called when loaded
the firmware for the part since the firmware image is so large and it
takes a long time to load. However, I might have a fix
Zaptel release.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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Ex Vito wrote:
On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
One thing also I would like to see is your kernel .config file. Another
thing that would for sure remove that warning is to disable the kernel
softlockup detector which is giving a false lockup
. That should disable the softlockup
detector.
Matthew Fredrickson
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.10
Zaptel Echo Canceller: MG2
ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154
wcte12xp: Setting up global serial parameters for T1
wcte12xp
Ex Vito wrote:
On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
One thing you can also do is pass the nosoftlockup kernel parameter
into the kernel from the bootloader. That should disable the softlockup
detector.
Tested with no 4K stack kernel
it solved
Please contact technical support. You need to get the new version of
the firmware for that card, and they will be able to give it to you.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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have, you may have some other issue
that you are dealing with.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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revision of the firmware. Please
contact technical support and they should be able to get it to you.
Matthew Fredrickson
thanks
mike
This E-mail, including any attachments, may be intended solely for
the personal and confidential use of the sender and recipient(s) named
options like echocancel and
echocancelwhenbridged apply the same to hardware and software echo
cancellers.
Matthew Fredrickson
Digium, Inc.
know automatically to use the hw ec rather than the software one?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
technical support. They will be able to help you
with whatever issue you may have.
Matthew Fredrickson
The URL provided does not contain firmware for the VPMADT032
I* have logged a query with digum. Is there a URL to get this firmware from?
On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora
gets
clipped / dropped when you speak.
Please contact Digium technical support about this. This is definitely
something that we need to work with the vendor of the echo canceller IP
about.
Matthew Fredrickson
Thanks
Ruben
Lex Lethol escribió:
Hi,
I've used all kinds of digium
it resolved as soon as possible :-) !
Matthew Fredrickson
Digium, Inc.
zaptel.conf
# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must
on the matter, the IAXy does indeed
have some echo cancellation built in. It has to since it interacts with
a phone via a 2 wire to 4 wire conversion with a hybrid.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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is for analog lines, IIRC.
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Software/Firmware Engineer
Digium, Inc.
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this might
be happening. We'll keep you posted when we find out what's wrong.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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you recompile them in the correct order (Zaptel
1st, then libpri, then Asterisk)?
Matthew Fredrickson
[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Unlinking slave 1 from 47
[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 12 from conference 9/47
[Jan 16 09:18:41] DEBUG[10183
of coefficient parameters in dB.
In order to use the new analysis calculations, you do not need to pass
any sort of special parameters to fxotune, it does the new analysis
technique by default.
Please let me know if you have any issues as well. Thanks!
--
Matthew Fredrickson
Software
with the purchase of the card. I haven't heard of
anything like this, although posting your kernel panic output would
help. But it would be best to handle this through tech support.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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. For digital T1/E1 cards, the only way to do it is with the gain
options in zapata.conf.
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Software/Firmware Engineer
Digium, Inc.
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and others have been working on it and has improved its
reliability to the point of fixing most if not all of the previously
outstanding issues. I recommend trying it again.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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possible?
Hold that thought just for a little bit :-)
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Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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Matthew Fredrickson
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Digium, Inc
is saying RLT not allowed?
Also, you have to make sure that it is between an inbound call (to
Asterisk, from DMS) and an outbound call (from Asterisk to DMS). It
should already check for this in libpri, but I figured I'd mention it
just to be sure.
--
Matthew Fredrickson
Software/Firmware
within a kernel driver, which is not a pretty way to do
what they are trying to do.
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Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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when layer 2 is not up, constant red when layer 2 is up, and it will
flash green when D-channel messages are sent on the port.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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protocol on the PRI. Asterisk
does not support this mode.
It may work with either the 4E or 5E switchtypes. There is some code
that (I didn't write it) I think unbusies the channels that is executed
for one or the other switchtypes.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium
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