Re: [Asterisk-Users] Problems connecting xlite phone

2004-07-28 Thread Jason A. Pattie
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Geoff Nordli wrote:
| How simple it is to kiss a couple of days away over something really
minute.
|
| It was definitely a client configuration issue.  I configured Proxy 1 to
| attach to asterisk.  I really needed to configure [Default].  Once I
| configured Default then I was off to the races.
Or I think you might be able to right-click on the interface (which
doesn't work under Wine, last time I checked; it locks up the interface)
and choose the account you want to use.
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Re: [Asterisk-Users] Echo on a PRI

2004-07-20 Thread Jason A. Pattie
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Andrew Kohlsmith wrote:
~  The only problem with echotraining=yes is the 8/10s delay before
audio is
| heard -- sometimes the start of conversations is cut off since our
| receptionist doesn't have to life a receiver from the cradle to her ear.
I normally wait about a second after I pick up the phone until I hear a
very small click.  I think that might be the end of the training period.
~ Then I proceed with my introduction.  It seems to work quite well.
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Re: [Asterisk-Users] Sort of OT: Recommended USB handset for use with iaxComm?

2004-07-13 Thread Jason A. Pattie
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Nate Carlson wrote:
| On Mon, 12 Jul 2004, Brian Weaver wrote:
|
|Tell him to spend $70 on a one port ATA like the Sipura-1000 , a USB
|headset will run $40-60.  For a little more money, you'll have an
|external box that is not leashing you to a computer or one location.
|
|
| Actually, he'd like the USB handset over the ATA - he's not sure what type
| of 'net access he's going to have at places, and doesn't want to have to
| deal with provided an ethernet connection to the ATA.
I have been successfully using the Plantronics DSP-400 USB headset with
both gnophone and iaxComm.  gnophone audio is beautiful and is
apparently a very good client for recovering from stray bit loss, etc.
However, it is no longer supported, even in Asterisk out-of-the-box.
You have to compile IAX version 1 support back in if you want to use
gnophone.  iaxComm is still, in my opinion, quite rough around the
edges.  The audio quality on transmission to another end is very
scratchy and poppy.  I'm not sure if this is a function of CPU power or
the audio system that iaxComm uses.  I've been attempting to use it on
an iPAQ device running GPE.  I sucessfully compiled it and have it
running, but the audio is especially terrible on that device (H3670,
206MHz ARM, 64MiB RAM).  I was able to apply the patch that ziaxPhone
has for the iax2-parser not being word aligned, and that helped the
audio tremendously.  But it's still very poppy, echoy, feedbacky, and
scratchy.  Sort of like lurching through the audio stream.  I've learned
that if I turn the micrphone down to almost nothing it works a lot
better for awhile.  After awhile, though (30 to 45 sec.), the audio
output almost fades to nothing on the iPAQ device.
Anyway.  You might take a look at the USB device available from
www.virbiage.com.  It might work with more than just their FireFly
client, especially if it shows up as an audio device under Linux.
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Re: [Asterisk-Users] FXO impedance matching

2004-06-23 Thread Jason A. Pattie
Robert Hajime Lanning wrote:
Echo echo ech ech ec ec e e . .
:)
quote who=[EMAIL PROTECTED]
What's the importance of the impedance matching in a FXO interface ?
If impedance matching is that important, then how is it accomplished? 
I'm fairly sure our X101P is not impedance matching properly.  I've 
never not had echo, and I've followed all the procedures in the wiki and 
from help that I've gotten from IRC.  The only way to hear anything is 
by turning up the gain in /etc/asterisk/zapata.conf, and I've been told 
this is a big no-no, as it increases echo problems.  But without it, the 
line is so low that it's useless.  However, if you plug a handset 
directly into the line coming from the CO, it's at a decent volume 
level, etc.  No echo, of course.

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Re: [Asterisk-Users] Any echo issues with phones from TDM400P X100P

2004-06-22 Thread Jason A. Pattie
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Rich Adamson wrote:
|Hi,
|I'm thinking of purchasing a TDM400P card and was wondering if anyone has
|experienced any echo issues with phones off these cards connecting to the
|PSTN via the X100P cards?
|
|I have had my fingers burnt with a Voip phone  X100P.
|
|
| I replaced two x100p's with a tdm04b 4-port fxo card, and the echo problem
| with pstn lines are absolutely no different; echo is still there and I've
| spent a substantial amount of time attempting to improve it (as an ex-
| telco engineer). Three pstn lines from two central offices all identical.
| Attaching these same three pstn lines to a Mediatrix 1204 totally
| eliminated all echo, however the 1204 had other problems that caused
| me to return it to the reseller. Tried ever combination of echo
cancellation
| options in Make and *.h files that folks have suggested, problem is
| still there and most disturbing during the first 15 seconds of all calls.
|
| As a non-programmer and * user for over nine months, its my opinion
(right,
| or wrong) that past echo problems have had two sources; one from pstn
| mismatch problems that have been heavily discussed since about September
| 2003. The other source has not yet been found/diagnosed and seems to
| be system dependent (and its not irq related either). You either have
| it or you don't, and the few developers that have either the x100p or
| the tdm card installed, don't seem to have the problem. Therefore
diagnosing
| the root cause is still waiting on someone with the problem _and_ the
| programming skills.
I absolutely second this notion.
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Re: [Asterisk-Users] Any echo issues with phones from TDM400P X100P

2004-06-22 Thread Jason A. Pattie
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Kevin Walsh wrote:
~  You should experiment with your configuration settings for the FXO
| card.  Check your gain values etc.  If you don't have any luck with
| that then try changing your zconfig.h to switch on CONFIG_ZAPTEL_MMX.
| You can also play with the AGGRESSIVE_SUPPRESSOR switch (same file).
Whoah.  I've used the AGGRESSIVE_SUPPRESSOR switch, but what does
CONFIG_ZAPTEL_MMX do for you?
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Re: [Asterisk-Users] Need guides on setting up PDA on asterisk server

2004-06-21 Thread Jason A. Pattie
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Holger Schurig wrote:
|He says he will release the sourcecode when he gets to a stable
|working release.
|
|
| He said the same to me, almost a month ago before his holiday ...
|
| However, I confess that QtIAX didn't come forward during this month
either
| :-)
|
|
|
|Do you think your QtIAX client will run on a 206MHz StrongARM
|processor?
|
|
| Probably. Just keep in mind that my embedded target is using Qt 3.x, not
| Qt 2.x.
|
| The real question is if I can get rid of all floating point. Both my
| target (Intel PXA255, 400 MHz) and your (Intel StrongARM, 206 MHz) don't
| have a copro, so we need to do all of our math with either integers or
| fixed point arithmetic. As a step into this direction I deleted quite a
| bunch of the iaxclient stuff, e.g. most of the echo cancellation code. I
| even thought about getting rid of iaxclient completely and using libiax2
| directly.
I guess that's why my compilation of iaxComm against wxGTK on Intimate
chokes whenever I select any of the options (like echo cancellation,
noise reduction, etc.).  Otherwise, it almost works fine.  This is using
the ALIGN32 patch from Stephan Kauss.  Incoming audio is fine, but
spoken audio (through the iPAQ's microphone to the other side) is very
choppy/screechy.  I guess I could try turning the microphone down even
more (using the gpe-mixer it's currently as low as I could get the
slider without it being totally off) using setmixer which I installed
under Intimate.  If I have the mixer settings too high for the master
volume, I get an awful feedback loop going.  Don't know if the iPAQ has
any sort of builtin feedback suppression circuitry or not.
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Re: [Asterisk-Users] Need guides on setting up PDA on asterisk server

2004-06-18 Thread Jason A. Pattie
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Holger Schurig wrote:
|I should follow this up to accurately state that audio was not
|operational in my test calls from the PDA.  I have patched the
|iaxclient library with the changes available from ZiaxPhone that word
|align the IAX2 library on the ARM platform.  I haven't finished
|compiling a new binary to test with.
|
|
| If you want to even patch ZiaxPhone, you can't: there's no source.
|
| There is something similar at my homepage,
| http://www.holgerschurig.de/qtiax.html. It doesn't yet run on my PDA) and
| lacks a config file support, but it's all source code.
He says he will release the sourcecode when he gets to a stable
working release.
Do you think your QtIAX client will run on a 206MHz StrongARM processor?
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Re: [Asterisk-Users] Need guides on setting up PDA on asterisk server

2004-06-17 Thread Jason A. Pattie
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ng kar fei wrote:
| How did you do that? need to cross-compile it?
| --- Simon Brown [EMAIL PROTECTED] wrote:
|
|I have successfully used SJPhone on my iPAQ 5450
|with asterisk.
|
|Simon
|
|
|
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On
|Behalf Of ng kar fei
|Sent: Monday, 31 May 2004 18:50
|To: [EMAIL PROTECTED]
|Subject: [Asterisk-Users] Need guides on setting up
|PDA on asterisk server
|
|
|Can PDAs be used as softphones/clients on asterisk?
|what i wanted to do is to set up 2 PDAs as
|softphone(client) which allows
|them to communicate each other through asterisk
|server(desktop)
|
|devices i have:
|
|pda compaq model 3680
|pda sharp sl5500
|access point
|desktop(asterisk)
You might look into using iaxComm or ZiaxPhone (available from
http://iaxclient.sourceforge.net/).  I have successfully gotten iaxComm
to compile and work on an iPAQ H3670 running GPE/Familiar Linux in the
past.  ZiaxPhone is supposed to run on the Zaurus or Opie.
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Re: [Asterisk-Users] Disable IAX1 Registrations

2004-06-17 Thread Jason A. Pattie
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Rich Adamson wrote:
|I was just noticing that my asterisk box is trying to register with my
|providers on both IAX1 and IAX2.  They do register on IAX2, but it
seems like
|it keeps trying to register with IAX1 also.
|
|Is there a way to disable this behavior, or am I just misunderstanding
it.  I
|just want to make sure I'm not wasting processor time and bandwidth
trying to
|register IAX1 when it is already registered with IAX2.
|
|I am using Asterisk 0.7.2 and GnoPhone 0.2.4, so I believe I still
need to use
|my IAX1 channel for that.  Also, I did find a patch that is supposed
to help
|this problem, but I don't know what it does, or if it has already been
|implemented.  The link to it is here:
|http://lists.digium.com/pipermail/asterisk-users/2003-December/031226.html
|
|
| Insert:
|  noload = chan_iax.so
| in the /etc/asterisk/modules.conf file. The next time asterisk is
restarted
| it won't try to load the old iax stuff.
Which will promptly cause gnophone to quit working.  :)
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Re: [Asterisk-Users] Need guides on setting up PDA on asterisk server

2004-06-17 Thread Jason A. Pattie
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Jason A. Pattie wrote:
| You might look into using iaxComm or ZiaxPhone (available from
| http://iaxclient.sourceforge.net/).  I have successfully gotten iaxComm
| to compile and work on an iPAQ H3670 running GPE/Familiar Linux in the
| past.  ZiaxPhone is supposed to run on the Zaurus or Opie.
I should follow this up to accurately state that audio was not
operational in my test calls from the PDA.  I have patched the
iaxclient library with the changes available from ZiaxPhone that word
align the IAX2 library on the ARM platform.  I haven't finished
compiling a new binary to test with.
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Re: [Asterisk-Users] iax codec problem

2004-06-09 Thread Jason A. Pattie
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Adam Hart wrote:
| Jason A. Pattie wrote:
|
| |
| | One workaround is to use Firefly, but that may not be for everyone?
|
| True.  I almost got it working under Wine, though.  Kept dumping files
| into C:\.  Probably just means I don't have the necessary dependencies
| or Wine doesn't have the capabilities needed to run this app., yet.
| Something about not being able to get timing from threads seemed to be
| the big killer.
|
| fixme:thread:GetThreadTimes Cannot get kerneltime or usertime of other
| threads
| fixme:thread:NtQueryInformationThread info class 9 not supported yet
|
| Oh well.  It was worth a shot.  At least part of the interface shows up
| on the screen before Wine bombs.
|
|
| Lol, that's a decent attempt (funny thing is that's the callstack that's
| having that problem but I can port it already - just not the GUI) -
| We're currently looking at porting, looking at the various
| cross-platform windowing libraries. If you have any suggestions or
| information on porting a windows GUI C++ program, send me an email
As much as I dislike it, wxWindows seems to be one of the most
ubiquitous cross-platform APIs for such a thing.  I'm sure there are
others, but it is released under LGPL licensing so you can distribute
commercial projects based on it (I think that's the way LGPL works).
Since it is so cross-platform compatible and runs on just about
everything you can think of (think iaxComm, which is using wxWindows to
be able to, currently, run on Windows, Linux, and Mac OS X), I found it
to be slightly slow.  Some other developers I talked with said that any
additional windowing toolkit on top of the underlying windowing
components is going to make your application slower as far as the
interface is concerned.  I'm pretty sure they are addressing this,
though, in either the development version or later versions.
In particular, I was running iaxComm on GPE on Familiar Linux on the
iPAQ.  So, that platform is already slow to begin with, and putting
wxWindows (technically wxGTK, the version of wxWindows ported to the
GTK(+) libraries) on top of that seems to have slowed the application
down even more, and then on top of that the application being an IP
softphone app., and the processor without an FPU unit, etc. etc.  You
get the picture.  I'm attempting to get iaxComm compiled again with the
iax2_parse patch from the ZiaxPhone project which makes IAX2 word
aligned and fixes a major sound issue which I appear to be having, but I
haven't gotten wxGTK to successfully compile and iaxComm to compile
against it again.  Don't remember exactly what I did the first time.
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Re: [Asterisk-Users] iax codec problem

2004-06-08 Thread Jason A. Pattie
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Tor Houghton wrote:
| On Sun, Jun 06, 2004 at 04:25:32PM -0400, Tim Sailer wrote:
|
|On Tue, Jun 01, 2004 at 07:49:29PM -0500, Yelson Vivas wrote:
|
|Hi everybody
|
|i have a problem trying to connect an incomming phone call from pstn
to my
|(soft phone) iaxcomm, the  phone rings but when i try to answer the call,
|asterisk sends a message like this.
|
|Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping
|incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format
GSM since
|our native format has changed to ALAW
|
|I have the same problem. IAXCOMM works fine with * 0.7.2, but not 0.9.
|However, you can make calls fine, just not pick up inbound calls.
|
|
|
| One workaround is to use Firefly, but that may not be for everyone?
True.  I almost got it working under Wine, though.  Kept dumping files
into C:\.  Probably just means I don't have the necessary dependencies
or Wine doesn't have the capabilities needed to run this app., yet.
Something about not being able to get timing from threads seemed to be
the big killer.
fixme:thread:GetThreadTimes Cannot get kerneltime or usertime of other
threads
fixme:thread:NtQueryInformationThread info class 9 not supported yet
Oh well.  It was worth a shot.  At least part of the interface shows up
on the screen before Wine bombs.
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Re: [Asterisk-Users] 100 analog phones?? HOWTO?

2004-05-26 Thread Jason A. Pattie
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Steven Critchfield wrote:
|   I was also thinking that if I used more than one machine with TDMoE I
|could potentially have better performance.
|
|
| I still doen't understand why someone would do TDMoE when IAX provides
| more flexibility.
I agree.  I'd stick with IAX or IAX2 trunking, from personal experience.
~ Using TDMoE caused my machines to become unstable, reboot, kernel
oops/panic, etc.  It was not a pleasant experience.  And, you have to
make sure that you start and stop the *'s together.  I can't remember if
I ever got that working or if it always trashed the other box when I
tried to bring * down with TDMoE in place.
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Re: [Asterisk-Users] 100 analog phones?? HOWTO?

2004-05-25 Thread Jason A. Pattie
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Andrew Kohlsmith wrote:
|   Does anyone know the best approach to take for handling 100 analog
|phones?  It seems to me that a chassis like Carrier Access or Adtran
|would work.  The chassis would do much of the hard work of converting
|the analog sound to data.
|   Any recommendations on hardware for the chassis?
|
|
| I'm partial to the Adit600 -- small, excellent quality, hot-swappable
| everything, hot-config through the network and can handle 48 lines.
|
| I hear the Adtran 750 is very good, too.
Would it be insane to recommend plugging analog handsets into those
devices that convert the analog extension into either a SIP or IAX phone?
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Re: [Asterisk-Users] Terrible TICKING sound

2004-05-20 Thread Jason A. Pattie
-BEGIN PGP SIGNED MESSAGE-
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Anon wrote:
| On Thursday 13 May 2004 11:57 pm, Jason A. Pattie wrote:
|
|-BEGIN PGP SIGNED MESSAGE-
|Hash: SHA1
|
|Steven Critchfield wrote:
|| On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote:
||Our problem ended up not being with Asterisk or Digium hardware.  It was
||the analog cordless phone.  We simply have to live with it.  What
||happens is whenever a connection is established and the phone is
||off-hook, an LED on the base lights up in a blink blink . blink
||blink . etc. pattern.  Everytime the LED lights, a pulse is sent to
||the phone.  It's especially bad when both lines are in use, as the phone
||is a two-line capable device.  Then you've got double the pulsing.
||
||This may have nothing to do with your problem.  Just wanted to get it
||out there in case anyone else runs into it, too.
||
|| Sounds like your phone needs either a aux power source to power that
|| led, or possible a little modification to clip that LED.
||
|| I would make sure your cordless phone's power supply is within spec. If
|| it is, Maybe you might want to look into one of the other comments a
|| while back on the list about upping the power on the SLIC(?). You might
|| be able to provide enough power to the phone to not cause trouble when
|| it blinks the LED.
|
|Well, the phone is using the power supply that came in the box.  :)
|
| If the phone is old and had average or more use, the transformer in the
| wall-wart might be operating at less capacity than when it was new, and
| might not be adequate now.
Hmm.  It's fairly new.  Less than 2 years old.
| You can get a good, inexpensive replacement wall-wart from
www.jameco.com.
Neat website.
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Re: [Asterisk-Users] Softphone lag

2004-05-20 Thread Jason A. Pattie
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Jason Williams wrote:
| This is normal for all VoIP communication there is nothing to wory about
| and the lag is not heard in normal use.
|
| Jason
|
| At 13:50 20/05/2004 +0530, you wrote:
|
| Hi,
| IF i use a sip softphone or a iax softphone with asterisk, i get a lag
| of about 1 second.
| The two phones were on 2 different pc's near me. When I speak on one,
| i hear it on the other after about 1 second.
| I tried using iaxComm, Xten Xlite, etc. Same.
|
| FYI: The codec used was GSM.
Would it get better if a straight codec were used?  I.e., one that
does full 8KBps (I think that's either ALAW or ULAW).  In other words,
bandwidth usage would obviously increase over a codec like GSM, but
would the lag be reduced because no translation has to be done at all,
just send it to the audio device, no (de)compression, etc.?
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Re: [Asterisk-Users] 4 X100P + 1 TDMP400(4 FXS): Only by miracle?

2004-05-17 Thread Jason A. Pattie
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Isamar Maia wrote:
| Hi folks,
|
| I'm trying to make an * PBX for a customer using 4 X100Ps
| and 1 TDM400p(4FXS).
| The problem I'm facing is to make one unique IRQ for each
| PCI slot/board since shared IRQs create all kind of weird noises
| and echos.
| Anybody got any workaround for that?
| Any recommended motherboard to accomplish that ?
| Currently, I'm playing with an ASUS A7V600.
Have you looked at possibly using the TDM400P with 4 FXO modules?  Then
you would only need to have 2 cards (currently) in the system and
possibly have room for expansion in the future, if needed.
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Re: [Asterisk-Users] Terrible TICKING sound

2004-05-13 Thread Jason A. Pattie
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Steven Critchfield wrote:
| On Tue, 2004-05-11 at 17:33, Jason A. Pattie wrote:
|
|
|Our problem ended up not being with Asterisk or Digium hardware.  It was
|the analog cordless phone.  We simply have to live with it.  What
|happens is whenever a connection is established and the phone is
|off-hook, an LED on the base lights up in a blink blink . blink
|blink . etc. pattern.  Everytime the LED lights, a pulse is sent to
|the phone.  It's especially bad when both lines are in use, as the phone
|is a two-line capable device.  Then you've got double the pulsing.
|
|This may have nothing to do with your problem.  Just wanted to get it
|out there in case anyone else runs into it, too.
|
|
| Sounds like your phone needs either a aux power source to power that
| led, or possible a little modification to clip that LED.
|
| I would make sure your cordless phone's power supply is within spec. If
| it is, Maybe you might want to look into one of the other comments a
| while back on the list about upping the power on the SLIC(?). You might
| be able to provide enough power to the phone to not cause trouble when
| it blinks the LED.
Well, the phone is using the power supply that came in the box.  :)

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Re: [Asterisk-Users] Terrible TICKING sound

2004-05-11 Thread Jason A. Pattie
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Ryan Courtnage wrote:
|
| On 11-May-04, at 8:45 AM, Ryan Courtnage wrote:
|
| I've fought with this problem on and off.
|
| Number 1 thing to check is /proc/interrupts to ensure that your card
| isn't sharing an interrupt with something else.
|
| Number 2 is a bit of an unknown variable - my guess is either
| electrical noise, or perhaps vibrations affecting your card inside
| your box.  I find that carefully remounting my tdm400p/x100p so that
| nothing at all is touching it (no wires, no plastic, nothing - except
| at the mount point) will make the problem go away the majority of the
| time.  If it doesn't go away, try re-mounting again.
|
|
| Currently, I'm working with a tdm400p with 2 FXO, 2 FXS.  the persistent
| tick tick tick tick has come back (after 3 days tick-free operation).
|
| Has anyone else experienced a similar problem with ZAP channels?  What
| steps did you take to resolve it?
Our problem ended up not being with Asterisk or Digium hardware.  It was
the analog cordless phone.  We simply have to live with it.  What
happens is whenever a connection is established and the phone is
off-hook, an LED on the base lights up in a blink blink . blink
blink . etc. pattern.  Everytime the LED lights, a pulse is sent to
the phone.  It's especially bad when both lines are in use, as the phone
is a two-line capable device.  Then you've got double the pulsing.
This may have nothing to do with your problem.  Just wanted to get it
out there in case anyone else runs into it, too.
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[Asterisk-Users] Disabling agent call logging

2004-05-11 Thread Jason A. Pattie
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Anyone know how to stop agent calls from being logged?  It appears that
the options in /etc/asterisk/agents.conf do not work with respect to
call logging.  All calls are logged in
/var/spool/asterisk/monitor/something-agent#-something else.wav, even
with the option to log disabled.  It records 3 files for each call: an
incoming file, an outgoing file, and another file, which I assume is the
merging of incoming and outgoing data (haven't actually listened to any
of these files, yet).
The only reason I'm complaining is that it ate up all the hard drive
space on my first Asterisk system before I realized what was happening
(2.5GB free space ... gone).  I didn't see a need to keep the files, so
I've just been deleting them, but I'd like to not have them recorded in
the first place (or is this something that isn't legal?).
Thanks.

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Re: [Asterisk-Users] No Audio from Hard Phone to SIP

2004-05-07 Thread Jason A. Pattie
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Eric Wieling wrote:
| Allow ULAW or ALAW, not both, at least for trying to solve a problem.
What is the difference between these codecs?  Which is better?

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Re: [Asterisk-Users] Channel Bank?

2004-04-07 Thread Jason A. Pattie
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John Vogel wrote:
| Four or five analog lines can be done with a single computer so no channel
| bank is needed. If you need 6 or more than there is also the choice of
using
| two machines and IAX.
I assume you would be using 4 or 5 X101Ps (or similar).  The only
problem I have currently with our systems that are connected via IAX2
switching is echo.  We have a single incoming line to one machine with
an X101P in it.  We then have another machine with a TDM400P and a
single analog extension.  The machines are connected via IAX2.  Just
today, one of our clients complained that he heard himself echoing.
Usually we hear ourselves echoing but the person on the other end does
not hear themselves echoing in the configuration we have setup.  Haven't
tried putting both cards into the same computer yet, but that will take
a lot of fiddling, since the new machine has an nForce2 chipset and
likes to assign the same IRQ to lots of different things.
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Re: [Asterisk-Users] Re: X100P Echo was: USB Headsets (Plantronics DSP-400)

2004-04-01 Thread Jason A. Pattie
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Doug Meredith wrote:
| Jason A. Pattie [EMAIL PROTECTED] wrote:
|
|
|Is there any possibility to remove the turnaround leg or whatever its
|called at the X100P?  I'm just thinking of a scenario where none of the
|outgoing signal is ever introduced to the incoming circuit.  That way,
|the echo problem simply disappears.  Or is this not possible?  What if
|the X100P were modified to do this in hardware?  Go ahead and let it
|balance the circuit if that is necessary, but then not introduce any
|of the outgoing signal onto the incoming circuit.
|
|
| I believe that some high-end hardware does this.  The ADIT 600 claims
| to.  I wish that the X100P did, as we have three that we are unable to
| use due to echo.
I would think this would be a no-brainer.  But, then, maybe I don't
understand the complexities of the problem that needs to be solved in
order to do this.  I just know that there are lots of systems out there
that deal quite adequately with this problem and don't experience echo
issues (i.e., cell phone to PSTN connections from multiple companies, etc.).
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Re: [Asterisk-Users] X100P Echo was: USB Headsets (Plantronics DSP-400)

2004-03-31 Thread Jason A. Pattie
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Brent Franks wrote:
| Agreed.  It seems that one out of about every 40 calls has echo problems
| for the duration of the cell.  I also notice it a lot more when the call
| is incoming vs. outgoing.  But it will still occur on outgoing calls
| occasionally.
|
| Is there any sort of inline echo cancellers that can help with this
| problem?  Having 6 lines doesn't justify the cost of a T1 or PRI.
Is there any possibility to remove the turnaround leg or whatever its
called at the X100P?  I'm just thinking of a scenario where none of the
outgoing signal is ever introduced to the incoming circuit.  That way,
the echo problem simply disappears.  Or is this not possible?  What if
the X100P were modified to do this in hardware?  Go ahead and let it
balance the circuit if that is necessary, but then not introduce any
of the outgoing signal onto the incoming circuit.
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Re: [Asterisk-Users] X100P Echo was: USB Headsets (Plantronics DSP-400)

2004-03-17 Thread Jason A. Pattie
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[EMAIL PROTECTED] wrote:
| Hi,
| The echo problem is the X100P.  The hybrid is 'unbalanced',
| and basically what happens is that the outgoing sound signal
| comes right-on back as an incoming signal.  The reason you
| don't notice it using the TDM400P is that the incoming sound
| is completely 'in-sync' with you talking through the
| handset.  The signal is 'bridged' in the computer, and there
| is (basically) zero delay.  Now, when using the X-Lite, the
| softphone has to do 'some' processing to get the signal
| coming in to the wires (X100P) to your computer speakers.
| This processing introduces delay (latency).  Now, you here
| your voice (outgoing signal, being 'turned-around' at the
| X100P phoneline interface) coming back at you, but with a
| more noticable delay.  Hence the echo.  The same effect also
| occurs when using SIP hardphones connected to [*] and
| calling out over the X100P.
| The solution is to implement a 'hybrid balancing' function
| at the X100P interface.  Traditioinally for FXO this has
| been done in hardware (not flexible) or more recently in
| software using embedded DSP chips providing adaptive
| real-time hybrid balancing.  The DSP algorithms are not very
| complex, and could conceivibly be run on the [*] main
| processor.  I have not looked into the codebase, however.
I have been fighting this very thing from the moment we implemented our
system.  Our original goal was to connect to the Asterisk box through
VPN tunnel connections and access the X100P interface card to receive
and make incoming and outgoing calls from wherever we happened to be (at
home, in the office, etc.).  However, the echo situation is still not
satisfactory on software phones.  I've got the echo cancellation tuned
to what I think is the best it's going to get without major improvement
in the algorithms.  Would a realtime kernel help in this regard?
BTW, there are echo cancelling algorithms available in the zaptel code.

| For a system with only a few FXOs it should not be a real
| problem.
| Notice, that a T1 interface is digital, and you do not have
| this 'turn-around' issue.
| Without looing at the Plantronics design specs, there is no
| telling whether it would work.  In any case, the real answer
If you are wanting the Plantronics to do echo cancellation, it's
probably not going to do that except for noise cancelling at the
microphone input.  I doubt it will do the kind of echo cancellation you
are desiring on its own.  At least I don't see anything in the Windows
tray icon that would give me this functionality.
| is to fix the problem right at the X100P interface.

I agree, which is what the echo cancelling algorithms are for in the
zaptel drivers.  They just don't seem to work 100%, yet.
| Cheers,
| Willy
|
| - Original Message Follows -
|
|Hello all,
|
|I'm thinking about getting the Plantronics DSP-400 headset
|for use with Xlite softphone.  I currently have a analog
|headset that does NOT have a DSP on board, which gives me
|mediocre call quality and echo when talking to the PSTN
|thru my X100P card.  I have zero echo when talking thru my
|X100P on my cordless phones attached to the Digium
|TDM400P.
|
|Before I got spend the money I was wondering if others
|using USB headsets with a DSP and getting good results?
|My thought was thought by using a headset with a DSP on
|board the echo would go away?
|
|Any advice on which USB headset in general to use with a
|softphone?
I routinely use the Plantronics DSP 400 (mostly with IAX softphones),
even under Linux.  The latest version of ALSA 1.3c fixed all my
problems.  It now works wonderfully with gnophone even.  Now I can use
it under Linux, not just Windows anymore, which makes me very happy.
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Re: [Asterisk-Users] AGI - IVR - Time Clock

2004-01-05 Thread Jason A. Pattie
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PBX wrote:
| I wanted to post the beginings of my latest IVR Project for an automated
| Time Clock software.
Not to alarm you too much, but MCI WorldCom has a patent on this kind of
thing and is suing people that develop/implement/use these kinds of
systems.  I wasn't aware that an idea could be patented or even an
implementation, but basically, from what I saw they patented a flowchart
(!!).
Basically, check into the relevant patents for this kind of system.

Just giving you a heads up.

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Re: [Asterisk-Users] Echo cancellation

2003-12-01 Thread Jason A. Pattie
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Nicolas Gudino wrote:
| If the terminating tail circuit has cancelable echos and if the echo
| canceler is enabled, you will hear echo for the first few utterances and
| then it will die away. After a few seconds of speech, the echo should be
| gone or at least very quiet compared to the echo level at the beginning of
| the call. This is the signature of a working echo canceler. 
In my situation where we use VoIP softphones connecting to an X101P card
to the PSTN, the other end hears us just fine and there is no echo.
However, there is substantial echo on the VoIP clients.  I did one
extended test with X-Lite (or maybe it was DIAX, but I have the feeling
the performance would have been the same) where I was in an extended
conversation.  Echo was quite bad for the first portion of the call, but
after approximately 1 - 2 minutes, it progressively got better until
after that time, it was still there, but at a very controlled, curtailed
level.  The beginning and end of the echoed portions were chopped off
and the volume level of the echo was quite a bit lower than the audio
level of the two parties, to the point that it was no longer distracting
for the VoIP client to talk.  This was using the MARK2 with AGGRESSIVE
enabled suppressor.
|Headsets are particularly notorious for poor echo performance.. This
|is due to lack of acoustic isolation. Perhaps you could test using
|headphones and a mic.
What does this mean?  That the earphones are feeding back into the
wrap-around mic?  I have a USB Plantronics DSP 400 headset running under
ALSA sound system.  Is it feeding through the plastic parts and entering
the microphone sitting out near my mouth?
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Re: [Asterisk-Users] Feedback with X100P and SIP fwd.pulver

2003-11-24 Thread Jason A. Pattie
Brian West wrote:

Ya learn to search the archives.  This has been covered MANY MANY times.
 

And I still haven't gotten the echo to go away completely.  I usually 
just end up making it worse.  :)

bkw

On Sun, 23 Nov 2003, VoIP Fan wrote:

 

Hello:

I have installed *. I configured my SIP account and my X100P. But when I call from SIP or from PSTN. The SIP extension hear an echo voice of its conversation. Anyone can help me???

Thanks,

voipfan
   

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Re: [Asterisk-Users] Low Volume X100P

2003-11-19 Thread Jason A. Pattie
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Rich Adamson wrote:
|Is it enough to just unplug the phones? or do I need to actually remove
|all physical connectors/interfaces from the line and run the line
|directly from the DMARC to the X101P line in port?
|
|Don't piss around; connect the X101P directly to the demarc with NOTHING
|else connected.  If that works, connect the other extensions one at a
time
|until the problem comes back.
|
|
| Suggest using some very basic troubleshooting techniques...
Thanks.

| Unplug all the phones; did it help or not?

No.  It did not.  The signal seems to be clearer but still very low.
Depending on which level echocancel is set to and whether AGGRESSIVE
cancellation is enabled or not on MARK2 determines the level of static
present and echo on the VoIP side of the connection, but the caller from
a normal analog phone line to the X101P hears none of this.  The call is
very clear and legible and extremely good (when gnophone going through
the VPN doesn't glitch every second or two and cut off what the person
was saying; otherwise, what's coming from gnophone is exceptionally good
quality).
Wonder of wonders, the default settings when setting echocancel=yes
(which I read previously on this mailing list is 128) work the best.
Setting echocancel to 256 causes some sort of drastic feedback that just
about blows the SIP/IAX user's eardrums out.
| If not, disconnect the extra wiring; did that help or not?

This is going to be more difficult, I think.  I would have to undo the
wiring from two boxes that the DMARC is split out to inside the building
for this particular line.
| If you are real sure there is nothing more between your * machine and
| the demarc, then call the telco (or your favorite service person that
| has the tools and knowledge to actually test what's left).
|
| It might even be worth your while to simply move the * machine next
| to the demarc, and connect it directly to the demarc (with no other
cabling
| involved whatsoever) and eval the difference.
I think I will try this.

| Then, to help others understand the issues, report back to the list
exactly
| what fixed the problem (assuming something realistic solved the problem)
| so that others can gain an understanding of unusual things impacting this
| stuff.
Of course.

Thanks for your suggestions.  I hope this problem can get licked.
Otherwise, we basically just have a voicemail system, and not a very
useable one at that.
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Re: [Asterisk-Users] Low Volume X100P

2003-11-19 Thread Jason A. Pattie
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Rich Adamson wrote:
|| Unplug all the phones; did it help or not?
|
|No.  It did not.  The signal seems to be clearer but still very low.
|Depending on which level echocancel is set to and whether AGGRESSIVE
|cancellation is enabled or not on MARK2 determines the level of static
|present and echo on the VoIP side of the connection, but the caller from
|
|
| If you are fighting only an inbound transmission level problem, then
| trying changing
|  rxgain=0.0
|  (or txgain=0.0 for x100p outbound)
| but, I'm fairly certain a change in those parameters does not take effect
| until either a complete * restart (or maybe even a complete zaptel/system
| restart). Someone else on the list might know for sure.
|
| Obviously, if you crank the gain up, it will have a degrading effect on
| the echo canceler as well.
Right.  The rxgain and txgain are both set to 3.0.  It appears that one
only needs to shutdown and restart asterisk.  At least, I've heard
different effects when I do this without unloading and reloading the
wcfxo and zaptel modules.
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Re: [Asterisk-Users] I hate to do this but..

2003-11-18 Thread Jason A. Pattie
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Jason A. Pattie wrote:
| I have this exact same problem as well.  We have a scenario in which we
| are not using any analog extensions, just SIP and IAX software based
| phones (DIAX, X-Lite, gnophone, (trying to use) linphone, etc.) with a
| single incoming POTS line to an X101P card.  First, the audio level
| coming from/to the X101P card is unacceptable, unless rxgain/txgain are
| turned up.  It starts becoming usable around 6.0.  The desired audio
| level is reached when the gains are increased to between 10.0 and 13.0.
| At 13.0, though, the sound becomes almost too distorted to be of use,
| but it is at a very acceptable loudness level.
|
| As far as echo is concerned, there don't seem to be any problems on the
| side of the person calling into the system to the X101P card.  However,
| if they connect to a SIP/IAX extension, or the SIP/IAX extension makes
| an outgoing call through the X101P card, the echo on the SIP/IAX
| extension is horrendous.  Of all the echo suppressors I've tried, MARK3
| works the best for not blowing your eardrums out.  The problem is
| excrutiating when using MARK2 with AGGRESSIVE cancellation.  When this
| is enabled, and the rx/txgains are anywhere above 0.0 (even 0.5), the
| SIP/IAX client receives some sort of feedback or distortion that can
| only be described as an extremely loud screech or squelch that hurts
| very much.  And, it is rather explosive in nature.  It is partly solved
| by turning down the microphone volume and gain on the SIP/IAX client,
| but it does not stop the problem.  If the rx/txgains are set to 0.0,
| then the volume level coming from/through? the X101P is so low, that
| even turning the volume up all the way on the SIP/IAX client, it is
| extremely difficult to make out what the other party is saying (there is
| a bit of ambient sound level in the location we are working from,
| though).  The same happens if voicemail is left using the X101P.
| However, if voicemail is left using the SIP/IAX client, volume levels
| are very acceptable.
|
| I don't understand what is going wrong.  Do I have a bad X101P card?
| I've tried this card in 3 different machines, so far.  I will be trying
| it in a fourth shortly and let you know how it goes.
Well, I've now tried it in my fourth machine, a PII 266MHz, 128MB RAM
system.  It's slightly better, and I have tried it using MARK3 and MARK2
with AGGRESSIVE enabled.  It seems that it doesn't do bad things when
the rx/txgain is set to 3.0 with MARK2 and AGGRESSIVE, but there is
still a slight echo of the speaker of the VOIP extension (SIP/IAX).  The
echo isn't as bad, because every time they talk, the beginning of the
echo starts fine, but the end gets chopped off, so it's kind of a
stuttering echo, now, which is more usable than before.  I've also tried
multiple combinations of echocancel on and off, and
echocancelwhenbridged on and off as well as the full range of echocancel
being set to 32, 64, 128, and 256.  128 seems to work the best with 256
making the echo seeming to be slightly louder and more distorted.  The
audio input now seems to be at a reasonable level when listening to the
phone line, but it is still a bit soft.  However, the voicemail
recording is still so low as to be almost unintelligible.
Thanks.

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Re: [Asterisk-Users] Low Volume X100P

2003-11-18 Thread Jason A. Pattie
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Kevin wrote:
| Ive tried that, all it seems to do is distort the audio quality and add
| much more echo.  I see others have had this same problem and I was
| wondering if it was resolved or just living with it.
Currently, our Asterisk PBX is unplugged.  I'm going to run some tests
with all the analog phones that were sharing the X101P's line removed.
Is it enough to just unplug the phones? or do I need to actually remove
all physical connectors/interfaces from the line and run the line
directly from the DMARC to the X101P line in port?
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Re: [Asterisk-Users] I hate to do this but..

2003-11-14 Thread Jason A. Pattie
Anton L. Kapela wrote:

Jeremy McNamara said:

 

Flebay it and use the proceeds to support Digium by purchasing Zaptel
hardware.  http://www.digium.com/
   

I like the zaptel cards, I really do. The price per port is better
than other fxo/fxs's in almost all cases and the support I've received
from Digium is second to none. However, deploying several pbx's for
friends (and using the X100p's), I've come to find that the echo
cancellation to be sub-par when contrasted with other makers. I
realize ecan is a huge and complex issue, but it's central to really
convincing me to 'buy' more Digium hardware.
Such is the catch-22; it seems we need people to buy/support Digium in
hopes that they can improve things overall -- hoping this includes
work on dsp.c to achieve better ecan. It's difficult to recommend that
someone buy 4 or 6 ports of fxo knowing that they'll probably be
complaining to me about excessive echo.
Note: I've tried every possible echo suppressor listed in the
Makefile, and so far Mark2 + aggressive suppression works best, at the
sacrifice of FDX. I'd like FDX and aggressive ecan. G This isn't a
jab at zaptel or Mark, this note is to highlite one issue I've had
recommending Digium hardware.
 

I have this exact same problem as well.  We have a scenario in which we 
are not using any analog extensions, just SIP and IAX software based 
phones (DIAX, X-Lite, gnophone, (trying to use) linphone, etc.) with a 
single incoming POTS line to an X101P card.  First, the audio level 
coming from/to the X101P card is unacceptable, unless rxgain/txgain are 
turned up.  It starts becoming usable around 6.0.  The desired audio 
level is reached when the gains are increased to between 10.0 and 13.0.  
At 13.0, though, the sound becomes almost too distorted to be of use, 
but it is at a very acceptable loudness level.

As far as echo is concerned, there don't seem to be any problems on the 
side of the person calling into the system to the X101P card.  However, 
if they connect to a SIP/IAX extension, or the SIP/IAX extension makes 
an outgoing call through the X101P card, the echo on the SIP/IAX 
extension is horrendous.  Of all the echo suppressors I've tried, MARK3 
works the best for not blowing your eardrums out.  The problem is 
excrutiating when using MARK2 with AGGRESSIVE cancellation.  When this 
is enabled, and the rx/txgains are anywhere above 0.0 (even 0.5), the 
SIP/IAX client receives some sort of feedback or distortion that can 
only be described as an extremely loud screech or squelch that hurts 
very much.  And, it is rather explosive in nature.  It is partly solved 
by turning down the microphone volume and gain on the SIP/IAX client, 
but it does not stop the problem.  If the rx/txgains are set to 0.0, 
then the volume level coming from/through? the X101P is so low, that 
even turning the volume up all the way on the SIP/IAX client, it is 
extremely difficult to make out what the other party is saying (there is 
a bit of ambient sound level in the location we are working from, 
though).  The same happens if voicemail is left using the X101P.  
However, if voicemail is left using the SIP/IAX client, volume levels 
are very acceptable.

I don't understand what is going wrong.  Do I have a bad X101P card?  
I've tried this card in 3 different machines, so far.  I will be trying 
it in a fourth shortly and let you know how it goes.

Thanks.

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Re: [Asterisk-Users] IAX/IAX2 encryption?

2003-11-11 Thread Jason A. Pattie
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Chris Albertson wrote:
|I'd really like to see this.  Maybe each user could have a little
|black
|plastic key they could insert and turn to go secure.
|
|
| The PGP documentation suggestes that users cary their key
| in a floppy and never copy the key file to the hard disk.
| So your little black plastic key is a floppy with the write
| tab punched out.
|
| Where I work, our Air Force costomers are using a system
| where you need a smart card to make a PC work.  All  the
| PCs are basically dead unless you put your card in the slot.
|
| We have some Sun Rays in our lab with card readers.  You can
| be editing a text file.  Pull the card, the screen blanks
| when you come back from the restroom and find someone using
| your computer, no problem you just put the card in any
| random Sun Ray and the scren comes back with e editor up and
| the cursor where you left it.  These Sun Rays sell for only
| about $1K each.  Given time this kind of thing will be
| universal.  It's already cheap.
|
| At Sun, at least in one office programmers are not alowed to
| own computers.  Every time they come in the office, in the morning
| or after lunch they are asigned a random cubicle with a random
| workstation.  Forces then to walk the walk and not just talk it.
| with respect to paperless office.
|
| Last place I worked at the desktop PCs (except for the technical
| staff) where all diskless.  No disk = no configuration.
| Nothing to mess up.  They sucked up an rRAM disk image off the
| net at boot time.
Was this by chance using LTSP?  (www.ltsp.org)

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Re: [Asterisk-Users] XTEN-Lite Bad sound!

2003-11-03 Thread Jason A. Pattie
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WipeOut wrote:
| Has the bad quality started just recently? Has it ever worked nicely for
| you??
It depends.  :)

| If either of these is yes..
|
| What has changed in your setup? Have you recently upgraded to a  newer
| CVS??
I have a fairly recent version of Asterisk, but X-Lite has always worked
like a charm when accessing the PBX functions directly (i.e., leaving
voicemail, listening to menus, checking voicemail, etc.).  However,
whenever I receive or make a call through the X100P card, the people on
the other end say that I am garbled and the audio is very choppy and
echoy.  I also notice an on again, off again kind of beating in
the audio stream whenever I am making a call.  There is a period of
exactly 4 beats that seem to be more or less the same volume, very low
and in the background, that span approximately less than a 2 second
period.  Then there is a period of silence for less than 2 seconds, and
then the cycle is repeated.
| I don't have an answer for you but at least it may stop others falling
| into the same problem if somthing can be identified as the cause..
|
| later..
Thanks.

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Re: [Asterisk-Users] XTEN-Lite Bad sound!

2003-10-30 Thread Jason A. Pattie
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Ariel Batista wrote:
| Ok I have a question. I have Xten-lite working with our Asterisk
system and I am able to make and get calls.  But the main problem is the
sound is very choppy and sometimes it cuts off words.  I have tested it
with ulaw and alaw as well as GSM.  They all do the same.  ulaw seems to
work better.
I have exactly this same problem as well.  It's even worse when running
X-Lite under Wine under Linux.
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Re: [Asterisk-Users] IAX and Jitter problem

2003-10-08 Thread Jason A. Pattie
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Joe Dennick wrote:
| I'm coming at this thing from an Operational standpoint rather than a
development standpoint.  Viewing your problem from that angle, I wonder
how well your network is performing.  Could you have a cable problem
that the Asterisk server hasn't reported (Layer 1); or perhaps your *
Server is connecting at 100 megabit/half duplex, but the switch is
configured (or auto-detected) 100 megabit/full duplex (Layer 2); or
perhaps you have a bad port on your switch (Layer 1 or 2); or perhaps
there is a mis-typed subnet mask or default gateway somewhere between
the systems that hasn't been caught yet (Layer 3).  Is your Asterisk
server busy doing anything else that's tying up resources?
|
| You also menitoned that you haven't yet found a soft-phone.  X-Lite
(from www.eten.com) works really well on Windows workstations.
And Linux too! (using Wine).  There are a few gotchas that you have to
work around, but other than that, it works perfectly fine.  See my
previous posts about this topic.
Also, I think that should be http://www.xten.com.

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[Asterisk-Users] Nortel M Series phones support

2003-09-29 Thread Jason A. Pattie
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I've searched the mailing list quite extensively, but didn't come up
with anything promising (some things wer helpful, though).  Does anyone
know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310)
phones can be made to work with the TDM400P card or if they are ADSI
compatible at all?  I kind of doubt they will work if they are not
compatible, but I don't know what it would take to plug them directly
into a * box.
Thanks.

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[Asterisk-Users] X-Lite for Linux

2003-09-26 Thread Jason A. Pattie
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No, not really.  But X-Lite for Windows works just fine running under
Wine on Linux.
You have to change some of the auto detection features, specifically
auto detection of the IP address.  You must manually enter the IP
address that X-Lite will use, and then it works perfectly.
Now I have a decent cross-platform SIP softphone client.  :)

(now if only my laptop could record sound, I could really use it)

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Re: [Asterisk-Users] X-Lite for Linux

2003-09-26 Thread Jason A. Pattie
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Andrew Kohlsmith wrote:
|No, not really.  But X-Lite for Windows works just fine running under
|Wine on Linux.
|
|
| I thought gnophone and kphone both worked well?
They do, for what they do.

gnophone is (to my knowledge) and IAX-only client.  kphone (please
someone enlighten me) cannot inject DTMF digits into the stream, at
least I have not found a digit pad or similar way to do it.  I was
finally able to get linphone to work with * by having * trust the IP
address linphone was coming from.  linphone apparently cannot correctly
authenticate to * properly.  Don't know what's going on there.  linphone
does have the ability to inject DTMF digits and seems to work properly
from that respect.
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Re: [Asterisk-Users] X-Lite for Linux

2003-09-26 Thread Jason A. Pattie
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WipeOut wrote:
| Small problem with that theory, It look slike Xten have removed
| Transfer functionality from the X-Lite product..
And you only get 3 lines, where with the X-Pro version you get 6.

| Which then poses the question..
|
| Buy the X-Pro product for $50 and a fair quality headset for $20 (total
| $70) or buy a hardphone for $65 and be able to run it independently of
| the PC..
|
| Hmm.. Tuff choice!!
Good point.  Although, the more X-Pro licenses you purchase, the cheaper
they get.  Don't know if that's the same for hardphones.  But, why buy a
headset for the softphone?  You can do echo cancellation, right?
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Re: [Asterisk-Users] Having problems with S100U

2003-09-12 Thread Jason A. Pattie
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Jason A. Pattie wrote:
| Mark Spencer wrote:
| | Remember the S100U has to generate high voltage (-48V when idle, plus
| | about 70v a/c on top of that when ringing), and it does it all with 5V.
| | It does get a little warm.  That's normal.
|
| Wow.  I hadn't thought of that.  That's gotta put a drain on a USB
| system.  Imagine if you wanted to load an entire 127 extension phone
| bank off one USB port (daisy chained USB hubs) ...  Supposedly it's
| theoretically possible to connect a maximum of 127 USB devices to a USB
| port (or something like that).
Extending this thought further: has anyone thought of putting multiple
S100U's together into an enclusure (maybe Digium?) and selling it as a
USB channel bank?  It would maybe be a cheap alternative and increase
expandability (unless USB is not stable enough).
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[Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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I've tested the S100U unit in two different computers both with
different USB subsystems.  The one it worked the best in used the
usb-uhci USB driver.  The other system uses the usb-ohci USB driver.  To
make sure it was not the installation, I used the same hard drive that
had asterisk in both systems.  And I moved the X101P and S100U between
both systems to test.
I'm using debian testing with a custom compiled
vanilla+debianlogo+freeswan+preempt(with and without preempt still have
the exact same issues) 2.4.21 kernel.  I used the CVS version of * from
2003-09-08.
First, on the usb-uhci machine, if I attempt to use uhci, the moment
after the driver says:
usb.c: registered new driver wcusb
wcusb: wc_detect_device
Second exit
the machine locks up completely.  I cannot even ping it after that or
switch virtual terminals or anything.  Hard reset is necessary.  Not
even unplugging the S100U does anything.  And this is the machine the
S100U works the best on (out of the two I tested with).  When using
usb-uhci, the device is detected fine, but almost immediately and at a
periodic interval around 2-3 seconds thereafter, a message is dumped
onto the console and into the kernel logs:
usb-uhci.c: iso_find_start: gap in seamless isochronous scheduling

Right before this message is dumped to the screen (and logs), a glitch
will happen to the handset connected to the S100U with the effect
noticed being that no audio or dialtone or key presses or general
input/output to/from the handset is processed.  It's making it very
difficult to use the system.  Although, if I press the phone digits long
enough, I can eventually make it into the voice mail system.  But I have
an even harder time listening to the vm system since the voice prompts
keep getting cut off at key points (like: press ... listen to new ... 2
to switch ...oxes, etc.).  Of course, each time this happens, the above
message is spit out on the screen.
It's even worse on the usb-ohci machine.  Most of the time, the S100U
doesn't even give me dialtone.  * always sees the handset pickup and
hangup, but only extremely rarely do I get dialtone.  And sometimes,
usually right after I receive dialtone, if I hangup and then pickup the
handset again, I am greeted with what appears to be a very sad attempt
at dialtone, almost like a canary growling.  Very much lower pitched
than dialtone and slower.  It's almost like the system is running
dialtone through molasses.  So I switched back to the usb-uhci machine
because at least it's partly working.
Any thoughts?  Previous e-mails on this list that I found tend to make
me think the S100U may be starting to die.  But I hope it's just a
kernel or driver issue.
Thanks.

NOTE:  Dialing into the X101P on either system worked flawlessly.  I
don't think there is anything wrong with the X101P.
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Re: [Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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Jason A. Pattie wrote:
| It's even worse on the usb-ohci machine.  Most of the time, the S100U
| doesn't even give me dialtone.
Sorry, forgot to mention that the S100U device is detected and setup
correctly on this machine and no error messages are logged or spit out.
~ From the machine's point of view, the S100U seems to be working
correctly in that modprobe wcusb works properly and * sees the handset
go offhook and onhook, but almost never dialtone is produced at the handset.
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Re: [Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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Thanks WipeOut.  I agree.  I really hoped that wasn't the case, though.

WipeOut . wrote:
| S100U's are very flaky.. I am on my third one...
|
| The first one lasted about a month and then started clicking and
freaking which got worse and worse to the point where there was just
noise.. and the noise was all comming from the S100U..
|
| The second one started by dropping calls all the time and occasionally
not providing dial tone.. this too got worse until it almost never
provided dial tone and if it did it dropped the call after a few seconds..
|
| My recommendation is to get a new one ad see if it solves your problems..
|
| Later..
|
|
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|Jason A. Pattie wrote:
|| It's even worse on the usb-ohci machine.  Most of the time, the S100U
|| doesn't even give me dialtone.
|
|Sorry, forgot to mention that the S100U device is detected and setup
|correctly on this machine and no error messages are logged or spit out.
|~ From the machine's point of view, the S100U seems to be working
|correctly in that modprobe wcusb works properly and * sees the handset
|go offhook and onhook, but almost never dialtone is produced at the
handset.
|
|- --
|Jason A. Pattie
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Re: [Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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WipeOut . wrote:
| S100U's are very flaky.. I am on my third one...
|
| The first one lasted about a month and then started clicking and
freaking which got worse and worse to the point where there was just
noise.. and the noise was all comming from the S100U..
|
| The second one started by dropping calls all the time and occasionally
not providing dial tone.. this too got worse until it almost never
provided dial tone and if it did it dropped the call after a few seconds..
|
| My recommendation is to get a new one ad see if it solves your problems..
|
| Later..
I have another observation for you.  Do your S100U's get warm?  I've
left this one plugged into the USB port and the red LED is lit on the
unit even though the wcusb driver is not loaded at this time.  I noticed
that the unit gets warm whenever the driver is loaded.  I've left it
running now for around 2 - 3 hours and it is still quite cool to the
touch.  I will install the wcusb driver and see if the unit heats up.
Maybe all these issues that people have with these devices are heat related?
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Re: [Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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Jason A. Pattie wrote:
| I have another observation for you.  Do your S100U's get warm?  I've
| left this one plugged into the USB port and the red LED is lit on the
| unit even though the wcusb driver is not loaded at this time.  I noticed
| that the unit gets warm whenever the driver is loaded.  I've left it
| running now for around 2 - 3 hours and it is still quite cool to the
| touch.  I will install the wcusb driver and see if the unit heats up.
| Maybe all these issues that people have with these devices are heat
| related?
It's now almost 2 hours later and the device has been spitting out the
error messages documented in my earlier posts.  The S100U is it seems
now at a stabilized heat level.  It is not uncomfortably warm but
feels like a transformer you might have plugged into the wall.
It felt warmer fairly soon after I modprobe'd the wcusb driver, around
20 minutes.
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Re: [Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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Ernest W. Lessenger wrote:
| My S100U also gets quite warm. I haven't had any trouble with it though.
Thanks.  I'm just trying to figure out as many variables as I can before
sending it back.
| --Ernest
|
| At 01:31 PM 9/10/2003, you wrote:
|
| -BEGIN PGP SIGNED MESSAGE-
| Hash: SHA1
|
| Jason A. Pattie wrote:
| | I have another observation for you.  Do your S100U's get warm?  I've
| | left this one plugged into the USB port and the red LED is lit on the
| | unit even though the wcusb driver is not loaded at this time.  I
| noticed
| | that the unit gets warm whenever the driver is loaded.  I've left it
| | running now for around 2 - 3 hours and it is still quite cool to the
| | touch.  I will install the wcusb driver and see if the unit heats up.
| | Maybe all these issues that people have with these devices are heat
| | related?
|
| It's now almost 2 hours later and the device has been spitting out the
| error messages documented in my earlier posts.  The S100U is it seems
| now at a stabilized heat level.  It is not uncomfortably warm but
| feels like a transformer you might have plugged into the wall.
|
| It felt warmer fairly soon after I modprobe'd the wcusb driver, around
| 20 minutes.
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Jason A. Pattie
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Xperience, Inc. (http://www.xperienceinc.com)
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Re: [Asterisk-Users] Having problems with S100U

2003-09-10 Thread Jason A. Pattie
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Mark Spencer wrote:
| Remember the S100U has to generate high voltage (-48V when idle, plus
| about 70v a/c on top of that when ringing), and it does it all with 5V.
| It does get a little warm.  That's normal.
Wow.  I hadn't thought of that.  That's gotta put a drain on a USB
system.  Imagine if you wanted to load an entire 127 extension phone
bank off one USB port (daisy chained USB hubs) ...  Supposedly it's
theoretically possible to connect a maximum of 127 USB devices to a USB
port (or something like that).
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Jason A. Pattie
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Xperience, Inc. (http://www.xperienceinc.com)
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