then
there is a project at http://www.ecsl.cs.sunysb.edu/fir/ that implements
real time audio processing, leveraging a GPU as a coprocessor, that
would likely provide the majority of the framework for getting the
blocks of data in and out.
Kris Boutilier
Information Services Coordinator
Sunshine Coast
John Drapner by the looks of his nametag... Captain
Crunch according to Google... A lucky man at that moment in the minds of some on
this list, but perhaps best thought of as the ultimate pioneer of 'open
telephony'. ;-)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
knowing intimate
details of both your and the ITSPs systems, which is unlikely to be
disclosed.
Out of interest though; what codec were you using
successfully andfor how long, and what do they recommend you use
now?
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
. It would
behelpful if you and/or others in the same situation could take a moment
to evaluate the performance ofhttp://bugs.digium.com/view.php?id=5520against your
current solutions and add your feedback.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
From
.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle
Sent: Friday, October 28, 2005
11:23 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sipura
841 echo cancel
from
CVS-HEAD without aggressive cancellation before taking time to do any of
the above. It can be dropped into stable if needed by just copying it
(and the contents of the header file) over the top of the mec2 files.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast
There are none that Im aware of at
the moment the different echo cancellers just drop in as code
replacements at compile time, so theyre completely insulated from the rest
of the system, and vice versa.
Sounds like a good idea for a trivial
patch anyone care to contribute such for
?
Kris Boutilier [EMAIL PROTECTED] wrote:
On Tuesday 11 October 2005 11:49, alan wrote:
After solving the other low hanging fruit audio issues in our Asterisk
PBX, we are left with occasional cases of severe echo which we have not
found a solution for yet.
{clip}
Most
, but it works fine for me which is why I shared it.
The Zaptel echo can will be fixed so it performs predictably for everyone
eventually, but until then go with 3rd party T1 gear if you want it reliably
avoided.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional
at
all costs. Thus;
Question...
1. Do I need to go have a little chat to my ITSP
Yes. However, try to come armed with a good sized selection of PSTN numbers
that are generating problem echos so they have something to actually look at.
Hope that helps.
Kris Boutilier
Information Services
echo cancellation
on SIP/IAX translated calls.
:-)
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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being dialed from
PRI B channels as the resulting signal levels are not calibrated. Try
troubleshooting that one through the repair service...
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
configuration, much like mine, probably predates that configuration
directive entirely.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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FYI, this is a bug that has been patched in cvs-head - see
http://bugs.digium.com/view.php?id=4468
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Reuter
Sent: Thursday, October 13, 2005 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial
Watch the output of 'pri debug span 1' on the Asterisk server while placing the
call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andy Goss
Sent: Monday, October 10, 2005 5:58
(if not more so) to the zaptel software echo can code.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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by increasing the 'jitterbuffers' parameter in
zapata.conf
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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the problem (temporarily).
Are you running stable or CVS-head? If CVS, what date?
Just trying to fill in the blanks for the list - I've not heard of this kind of
problem before.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
out in the world it becomes more reasonable to simply
provision your own echo cancellation system facing the PSTN just incase.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Kohlsmith
Sent: Thursday, September 29, 2005 2:23 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Best echo canceller?
On Thursday 29 September 2005 17:04, Claudio Canseco
.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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,
assuming your signal is exposed on a T1 somewhere. If it's IP all the way for
you then you're really just down to the handset vendors as far as I know -
Asterisk doesn't currently offer any form of echo cancellation on the VoIP side.
Hope that helps.
Kris Boutilier
Information Systems Coordinator
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood
Sent: Friday, August 05, 2005 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is this echo problem down to IP Phone hardware?
Kris
research.
Everything has been working fine for quite a while now, though I've not tried a
very recent edition of head.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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congestion of that packet.
Hope that helps.
Kris Boutilier
Information Services Coordinator
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Search the wiki (voip-info.org) for 'tellabs'. There is comprehensive
documentation there. Email me privately if you need more information.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
- about
us$0.50/channel. Its far more economical than the hair replacement treatments
needed after trying to get zaptel/mec2 to behave... :-)
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
-Original Message-
From: [EMAIL PROTECTED
%20Hardware%20Echo%20Cancellers
for details.
Approximate price? About us$35 per 24 channel card - check eBay for latest
offerings.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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being done very late
at night... See if you can get the ticket escalated to the local switch
engineer to truly understand what they've tweaked.
If you find out what they did, be sure to post it.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
regarding installing a Tellabs 257x with Asterisk.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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qualification more useful.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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know if patlooptest is at all useful at
the moment as it's rather poorly documented.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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http
.
There is more background on my experience with the T100P popclick issue in
http://lists.digium.com/pipermail/asterisk-dev/2005-May/012432.html
Hope that helps.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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what
is happening and check if there is packet loose?
Depends on the channel type (sip, iax etc.) - each channel has it's own variety
of commented out very low level debugging. Use the source... ;-)
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
process actually works?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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-as-backend-for-mail clients using
IMAP, some other mail protocol or even a per-mail client custom plug-in as
you've suggested.
Just my can$0.02
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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in all
cases.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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in mec2_const.h (pay
particular attention to MIN_UPDATE_THRESH_I) or get busy studying the refered
to Texas Instruments whitepaper and then uncommenting MEC2_STATS and/or
MEC2_STATS_DETAILED.
Good luck, you have an unenviable problem.
:-)
Kris Boutilier
Information Services Coordinator
Sunshine Coast
with setpci before. Can you elaborate on the
use and purpose of this command?
See:
http://www-106.ibm.com/developerworks/library/l-hw2.html
Also, for more PCI latency timer specifics:
http://www.reric.net/linux/pci_latency.html
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional
-Original Message-
From: snacktime [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 12, 2005 5:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] UNREACHABLE messages
I get these on a consistant basis for most of the providers I have
the audio - small timing
defects on the T1 can usually be noticed by dialing into a milliwatt() target
on the problem box across that link. That said, perhaps the jitter is too
slight to hear on a 1004hz sign wave and needs a T1 analyzer to detect properly.
Kris Boutilier
Information Systems
-Original Message-
From: David [mailto:[EMAIL PROTECTED]
Sent: Friday, May 06, 2005 10:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] my_zt_write
Hello Guys,
Any idea what this means:
WARNING[2138]: chan_zap.c:4409 my_zt_write:
Disciplines: Token Bucket
Filter) and Chapter 15.9 (Cookbook - The Ultimate Traffic Conditioner: Low
Latency, Fast Up Downloads).
Also http://www.tldp.org/HOWTO/ADSL-Bandwidth-Management-HOWTO/index.html (ADSL
Bandwidth Management HOWTO) may provide useful knowledge.
Enjoy.
Kris Boutilier
-Original Message-
From: Dan Goscomb [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 1:48 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] PRI debug
Hi
I have a problem in that every time i try to dial a number i get the
error back that the number is
-Original Message-
From: Aza [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 04, 2005 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] TE410P on Dell 2650
I have a problem with a Dell 1850 and a TE410P card as do a
few others
sides of a working iax.conf
that uses this strategy to do codec selection? I'm guessing it's a really
simple issue that'll pop right out once I compare to a working config.
Thanks.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
parties from
seizing it simultaneously.
Google can provide far more detail.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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.
For the uninitiated, Request Tracker is an elegant and light weight GPL'ed
trouble ticketing system: http://www.bestpractical.com Great for managing all
those user interactions stemming from a new deployment of Asterisk.
:-)
Kris Boutilier
Information Systems Coordinatior
Sunshine Coast Regional
' codes concept Nortel use for Voicemail
and so on.
However, if you're only serving one unexpanded MICS KSU then ditching it may be
more realistic.
:-)
Kris Boutilier
Information Systems Coordinatior
Sunshine Coast Regional District
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.
Kris Boutilier
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-Original Message-
From: Christian Gerstner [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 23, 2005 4:18 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Any Software Echo Cancellation in Asterisk?
Hi,
short question: Is Echo Cancellation in Asterisk somewhere
-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 23, 2005 11:58 AM
To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Any Software Echo Cancellation in
Asterisk?
C F wrote:
short question:
for a recent thread titled
'Tweaking AGGRESSIVE_SUPPRESSOR'
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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-supressed as the signal cannot be
differentiated from the far end signal, thus getting your gains correct is an
important part of PSTN interfacing.
Hope that helps.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 22, 2005 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] X100P voicemail volume too low (quiet)
I'm running Asterisk 1.0.6
long distance phone bill to come
seperated out and grouped based on the originating number - hence Station Level
Billing.
Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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efficient somewhere resulting in a greater delay. If you haven't changed your
configuration anywhere then try comparing the output of 'show translation' on
both versions to verify the codecs are performing similarly.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
-Original Message-
From: Kris Boutilier
Sent: Wednesday, March 09, 2005 10:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Tweaking AGGRESSIVE_SUPPRESSOR
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent
The fact that your SIP people are hearing their own voice, but the inbound
caller is not is the correct behaviour for your 'echoless' digital termination.
If you were to tap the channel in the T1 leaving your premises you would find
that the echo is coming in from beyond your interfaces.
and so on.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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IN OUT, Receive IN OUT and +/-48vdc. Demonstrating an effective, successful
echo cancellation system would probably be enough to loosen the purse strings
enough to cough up for an 'offical' chassis.
I can post edge-connector pinouts and wiring instructions to the wiki if there
is interest.
Kris
noise) but upgrading to a newer 2572 64ms unit pretty much
took care of that too. Regardless, do not underestimate the damage unresolved
echo can do to the viability of a VoIP project.
Hope that helps.
Kris Boutilier
-Original Message-
From: Dennis Webb [mailto:[EMAIL PROTECTED]
Sent
This may be of some use to you on your quest:
http://bugs.digium.com/bug_view_page.php?bug_id=0002820
-Original Message-
From: Kris Boutilier
Sent: Tuesday, March 08, 2005 2:01 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
Discussion
Subject: RE
The originating PRI system passes the entire dialed number in the d-channel
setup frame, thus the concept of a wait time for additional digits is
meaningless. Progressive digit gathering implies that the signalling is
occuring 'in-band' as would be the case with DTMF signalling on analog
lines.
From the wiki page for cmd Macro:
Note that you cannot use any other extension than 's' to construct the macro
as control is returned to the calling context when the end of the 's,'
priorities is reached.
So, rather than using a macro - which is simply intended to be a replacement
for a
Issue a 'zap show channels' in the console. Each of your b-channels should
be listed, regardless of span. Issuing 'pri show span n' should show the
d-channel(s) associated with span n. I suspect your issue is because of the
order of the declarations in zaptel.conf. Instead try the form:
the endpoints directly to each other. In this case, it means the
call will be taken off of the network and bridged internally to *b.
Hope that helps.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
-Original Message-
From: Sean Kennedy [SMTP:[EMAIL
when compiling the kernel. You'll see it as 100 on some
kernels and 1024 on others (particularly Redhat).
See: http://www.linuxgazette.com/node/view/8993
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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the call is inside or beyond the serving CO then it's all
inside a digital domain and there is no loss. If you should end up on a call
where the far end analog circuit has signficant loss then at least your
amplitude will be comparable to everone elses.
Hope that helps.
Kris Boutilier
Information
it go cancel fine on about 90% of my calls, but there
are some which simply defy cancellation. In each of these cases it's 100%
repeatable and seems to be something to do with the remote parties line or
specific telephone characteristics.
Hope that helps.
Kris Boutilier
Information Systems
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: November 5, 2004 12:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Adjusting txgain/rxgain
Hi,
On Fri, 2004-11-05 at 21:39, Kris Boutilier wrote
-Original Message-
From: Christopher Jacob [mailto:[EMAIL PROTECTED]
Sent: October 27, 2004 8:56 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Remote Voicemail
Hey all,
I have a bit of a conundrum I need some help with...
I have two servers. local and remote... The
Subject pretty much says it all...
Thanks.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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on order) however
is it possible to insert another can into the system somehow? For example,
if I were to run TDMoE to a second box and access to the t100p/channel bank
through there?
Any suggestions welcome (apart from curing the sidetone) :-)
Kris Boutilier
Information Systems Coordinator
Sunshine
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent: October 12, 2004 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Chaining more than one zap echo
canceller?
- I am very close to my serving CO which brings the line
12, 2004, at 10:26 AM, Kris Boutilier wrote:
I have Asterisk connected to a channel bank via a t100p card. There
excessive sidetone generated on the analog side due to an impedance
mismatch
- I am very close to my serving CO which brings the line
down to about
150ohms and the channel
To be clear then, my subscriber loop may have a DC resistance of 150ohms due
to the physical loop length but should have an AC impedance of 600ohm (+/- a
few)? I have to assume that as the telco cable guy was performing the test
he's telling me the AC impedance.
If so, then would I need to insert
There is a bounty out for this feature. It's called the Simple Message Desk
Indicator (SMDI) protocol. See
http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+SMDI
-Original Message-
From: Clay Zevely [mailto:[EMAIL PROTECTED]
Sent: October 12, 2004 5:25 PM
To: [EMAIL
I was having this thought also and I couldn't find any implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt system and a few minor hacks on
app_voicemail.c
-Original Message-
From: Dominique Kull
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail forward to a remote server?
Why not just use rsync or netcat? There are about a dozen different
ways to do this.
John
Kris Boutilier wrote:
I was having this thought also and I couldn't find any
should one use this a 0dbm test source with ztmonitor? Am I
correct in understanding that a 0dbm level should provide a 100% drive in
the level monitor, not a 50% drive as is otherwise 'optimum' for regular
voice traffic?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
but it is very well
suited to real time discussions of problems...
That said, does anyone maintain a Googleable log of #asterisk?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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[EMAIL
the textual ID in the 'Display' element... However I understand
from http://resource.intel.com/telecom/support/tnotes/tnbyos/2000/tn033.htm
that there is no definitive standard for transmitting the name.
So, should even I be expecting Ast to put the name on the wire when it's
originating?
Kris
All good information, thanks. However this is private network between
Asterisk and a Norstar MICS about six feet away. So I'm holding both ends of
the link.
:-)
-Original Message-
From: David Troy [mailto:[EMAIL PROTECTED]
Sent: September 16, 2004 4:57 AM
To: Asterisk Users Mailing
-Original Message-
From: Jason Kawakami [mailto:[EMAIL PROTECTED]
Sent: September 13, 2004 9:08 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Astersk as AVAYA IVR
{clip}
Your assumptions on routing are correct. a path out of the
Index to the *
will require a path
The 'parkedcalls' code dynamically creates and deletes entries in the
dialplan to handle the calls that have been parked, so the parking lot must
not overlap your regular extensions. The initial parking extension is
statically created on startup, thus the 'exten =' entry is matching the
parking
and meet
their facilities requests, you'll get much more bang for your buck.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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-HEAD-08/13/04-10:37:13'.
Kris Boutilier
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Sunshine Coast Regional District
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?
On Tue, 7 Sep 2004 16:26:24 -0700, Kris Boutilier
[EMAIL PROTECTED] wrote:
I'm having a problem with intersite calls over IAX2 being abruptly
terminated. Nothing odd shows in any of the logs for
Asterisk or the host.
The only think I can think it might be is a lag-spike on
the site to site
At the moment we're not - the email notification from Comedian Mail has
been mostly sufficient. I do however have some Dialogic D/42-NS PBX
emulation cards and the plan is to use them to set and unset the MWI lamps
based on events pushed out of Asterisk.
They may be obsolete hardware but they
w/oji tterbuffer enabled?
On Tue, 7 Sep 2004, Kris Boutilier wrote:
{clip}
Run Asterisk with debugging turned on - see my various posts here
explaining how to capture it all in /var/log/asterisk/debug.
That will reveal all.
But I suspect you got some NAT between the end points
that helps.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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anything into the dial sequence. Can anyone
suggest a method of achieving this from within the dialplan rather than
modifying the Dial() application itself?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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'
provides:
Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
JitBuf Format
10.0.40.140 astpbx-woo 2/2 5/6 00040ms 0036ms
ms GSM
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
?
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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far today. I've had
reports of about 30 dropped calls.
What I really need is to find a better way to post-mortem debugging for
large numbers of concurrent IAX2 calls...
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
Unfortunatly no on both counts.
The arrangement right now has:
PSTN Trunks Stations - Nortel Norstar#1 -CT1- Asterisk#1 -IAX2-
Asterisk#2 -CT1- Nortel Nortstar#2 - Stations
The Asterisk boxes provide Voicemail to their sites Norstars and intersite
calls over IAX. Local Voicemail works
Take a look through the bounty offered at
http://www.voip-info.org/wiki-Asterisk+bounty+non-Bellcore-CLID . There may
be something very close to your requirements.
-Original Message-
From: Renato Mintz [mailto:[EMAIL PROTECTED]
Sent: September 7, 2004 6:01 PM
To: [EMAIL PROTECTED]
-Original Message-
From: Adam Goryachev [mailto:[EMAIL PROTECTED]
Sent: September 7, 2004 8:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Monitored outbound dialing via Zap
interface?
{clip}
Have you considered adding the r option
I trust you restarted (rather than just reloaded) after changing the
Voicemail configuration?
-Original Message-
From: box100 [mailto:[EMAIL PROTECTED]
Sent: September 7, 2004 8:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Problems with length of voicemail
I wonder if anyone
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