>>>>> "Gerald" == Gerald Henriksen <[EMAIL PROTECTED]> writes:
Gerald> On Thu, 02 Oct 2003 11:26:56 -0700, Jan Rychter
Gerald> <[EMAIL PROTECTED]> wrote:
>> Having worked with GPL software quite a bit, also in the commercial
>> world,
> "Brian" == Brian Schrock <[EMAIL PROTECTED]> writes:
Brian> Hello, I resolved my echo issue using grandstream/estara etc etc
Brian> sip phones and wcfxo interfaces from digium. I swapped out my
Brian> via kt400 based msi kt4vl motherboard for an asus p4pe? i845?
Brian> based motherboard
> "Ed" == Ed Dack <[EMAIL PROTECTED]>:
Ed> I've got * up and running everything seems to work ok except for
Ed> when you dial out using the X100P card.
Ed> Everything sounds great this end but the person you call complains
Ed> that they can't hear you very well (Very Whispered).
Ed> Is t
-patent" clause?
> > Uriel
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter
> > Sent: Thursday, October 02, 2003 2:27 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Us
>>>>> "Steve" == Steve Meyers <[EMAIL PROTECTED]> writes:
Steve> On Thu, 2003-10-02 at 12:04, Jan Rychter wrote:
>> I'm also hearing this, with an analog phone (connected to an
>> S100U). Rather annoying.
>>
>> Incoming calls
> "Eric" == Eric Wieling <[EMAIL PROTECTED]> writes:
Eric> Check /proc/interrupts to make sure the cards are not shareing
Eric> IRQs with anything.
Is there anything that can be done so that this is not a requirement?
There are (many) setups where this is simply not possible.
Other cards c
> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes:
[...]
Mark> No problem, it's easy to get confused :) I would, however, take
Mark> issue with the GPL being "evil". It's not my *ideal* license,
Mark> but it certainly is good enough.
Just for the reference, while we're at it. GPL does ha
> "Brian" == Brian West <[EMAIL PROTECTED]> writes:
Brian> Just a heads up.. you can't loop switch statements ie
Brian> BOX A switch => BOX B BOX B switch => BOX A
[...]
I was actually wondering -- why?
This is something I very naturally wanted to do the first time I
configured two *'s. I
> "LDM" == Louis-David Mitterrand <[EMAIL PROTECTED]> writes:
LDM> Having purchased a license for 5 g729 channels on Digium's web
LDM> shop I thought registration and installation would be a snap. NOT.
LDM> I followed registration instructions to the letter but it failed
LDM> with that mes
> "Shaun" == Shaun Ewing <[EMAIL PROTECTED]> writes:
Shaun> - Original Message -
Shaun> From: Chad R. Graham
>> For the first 15 seconds of a call I get echo on the ata 186 side
>> only. I assume after that the echo canceller kicks in but is there
>> any way to make it happen fas
> "Sean" == Sean P Robertson <[EMAIL PROTECTED]> writes:
Sean> I have. Heads up on the built-in sound. Like everything else on
Sean> the motherboard, it uses a VIA chipset and chan_oss will not work
Sean> with it.
Sean> Several posts have been made to the list in the past about the
Sean
> "Steven" == Steven Critchfield <[EMAIL PROTECTED]> writes:
Steven> On Wed, 2003-09-24 at 13:13, Jon Pounder wrote:
>> speaking of VIA - has anyone on the list looked at or used these ?
>> http://www.mini-itx.com/store/default.asp?c=2¤cy=2
>>
>> various collection of via based boards and
> "Mark" == Mark Spencer <[EMAIL PROTECTED]> writes:
Mark> I wouldn't mess with the gains if I were you. Mark
What do you mean?
Are the gains an unsupported feature? Aren't we supposed to adjust them?
I have some people who complain that they can't hear me when I dial out
using the X100P a
>>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes:
Jan> I'm having trouble with a WX100USB adapter and a Siemens Gigaset
Jan> cordless phone.
Jan> If I select fxols as a signalling method, calls are being
Jan> disconnected. Usually after ab
I'm having trouble with a WX100USB adapter and a Siemens Gigaset
cordless phone.
If I select fxols as a signalling method, calls are being
disconnected. Usually after about 4 minutes, and asterisk just says that
the phone has hung up.
If I choose fxogs, I immediately get a LINE IN USE message on
> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]>:
Jeremy> What part of "IN OTHER WORDS: Run Open H.323 v1.11.7, nothing
Jeremy> newer, nothing older if u want this to work." don't you
Jeremy> understand?
Well, I was trying to find out (politely) about some things. Please
allow me to paste
I have hit a problem where chan_h323 sometimes doesn't hang up properly
and stays stuck in the "Up" state, with asterisk consuming 100% of CPU:
*CLI> show channels
Channel (ContextExtensionPri ) State Appl. Data
H323/ip$127.0.0.1:30008/21552 (local 123
I've tried making calls using the console (both ALSA and OSS). ALSA
seems to work after applying the little fix posted on this list some
time ago by someone (which I'll submit into the bug tracker), but all I
get is one-way audio: I can hear the other end, but nothing gets
transmitted.
At first I
> "Steve" == Steve Underwood <[EMAIL PROTECTED]>:
Steve> Kim C. Callis wrote:
>> I was reading on www.vovida.org/applications/downloads/G729A/ (home
>> of VOCAL) pages, and that there is a free license use for
>> non-commercial for G.729A. Is that usable under Asterisk or strictly
>> a Vov
> "Mark" == Mark Spencer <[EMAIL PROTECTED]>:
>> This Windows binary is probably fairly easy to convert for someone
>> with sufficient skills. It's a simple library, COFF format. It's
>> probably sufficient to split it into .o files (using ar), then
>> convert the .o files (using objcopy --
When connecting an analog phone (Siemens Gigaset) to * via a WX100USB,
the phone displays "Out of area" first, and then the caller id. The two
displays alternate, making the caller-id hard to see.
Is there any way I can tell the phone to just display the caller id? Out
of area is a flag that gets
>>>>> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]>:
Jeremy> Jan Rychter wrote:
>> Please try to find a better solution.
>>
Jeremy> The DSP Group owns G.729. There is nothing anyone can do, they
Jeremy> have us by the family jewels. We
Seeing that many people here hit problems with activating their G.729
licenses, I decided to post my opinion.
I have purchased two G.729 licenses, for my private use. I did this even
though VoiceAge makes G.729 free for private use, as Windows
libraries. I guess a sufficiently motivated person cou
> "John" == John Todd <[EMAIL PROTECTED]>:
> "John" == John Todd <[EMAIL PROTECTED]> writes:
> What is the state of speex support in asterisk? I saw the codec seems
> to be there.
John> Install the Speex library support, and re-compile Asterisk.
John> There's probably a pre-compiled versi
> "Steven" == Steven Critchfield <[EMAIL PROTECTED]> writes:
Steven> On Fri, 2003-07-18 at 09:02, Chris Earle (CBL) wrote:
>> Agh
>>
>> I hate trying to sift through all these messages and keep track of
>> the various threads going on .
>>
>> Who else on here prefers the news
> "John" == John Todd <[EMAIL PROTECTED]> writes:
>> What is the state of speex support in asterisk? I saw the codec
>> seems to be there.
John> Install the Speex library support, and re-compile Asterisk.
John> There's probably a pre-compiled version of Speex for your system;
John> look a
What is the state of speex support in asterisk? I saw the codec seems to
be there.
Can speex be used on IAX2 links? Is there much work still to be done?
many thanks,
--J.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/
ing -- this has indeed helped and the quality is better, too!
But doesn't this mean I'm in trouble whenever the network decides to
order packets around?
--J.
Matteo> Il mer, 2003-07-16 alle 20:45, Jan Rychter ha scritto:
>> Hi,
>>
>> I'm running aster
Hi,
I'm running asterisk in the following setup
Phone -> WX100USB -> * -> Internet -> * -> WX100P -> PSTN
The two Asterisks talk to each other via IAX2 and use GSM for voice.
This seems to work quite well except for occasional pauses in voice
transmission. These seem to occur in _one_ direction
Does G.729 provide better voice quality than GSM?
(a question for people who have tried both)
--J.
pgp0.pgp
Description: PGP signature
>>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes:
>>>>> "John" == John Todd <[EMAIL PROTECTED]> writes:
John> For what it's worth, I have noticed the same problem, but I think
John> the problem is in IAX2, since
> "Matthew" == Matthew Hardeman <[EMAIL PROTECTED]> writes:
Matthew> I'm not familiar with the codec support in Gnomeeting, but
Matthew> have you tried a codec like iLBC? I had great success running
Matthew> ilbc over IAX2 between my home and office.
It doesn't really matter all that much
> "Jeremy" == Jeremy McNamara <[EMAIL PROTECTED]> writes:
Jeremy> Not after you've wasted the kind of money I did on that junk.
Jeremy> I was even stupid enough to pay the extra $30 per card for
Jeremy> G.729 and when I couldn't make it work on Linux, they told me
Jeremy> it would never wor
> "Matthew" == Matthew Hardeman <[EMAIL PROTECTED]> writes:
Matthew> Missing something? No...
Matthew> So far as I'm aware there is no freely available G729 codec
Matthew> available that will run under Linux... Kind of funny that
Matthew> there *is* one for Windows, isn't it?
Yes, puzzl
oticing these problems? Or are
>> people simply not using asterisk for VoIP connectivity over
>> wide-area networks this way?
>>
>> Or does it go away with g729 or other proprietary codecs?
>>
>> --J.
>>
> "Jan" == Jan Rychter <[EMAIL P
I'm curious. Isn't anyone else noticing these problems? Or are people
simply not using asterisk for VoIP connectivity over wide-area networks
this way?
Or does it go away with g729 or other proprietary codecs?
--J.
> >>>>> "Jan" == Jan Rychter <[EMAIL P
Hi,
I'm looking for a good codec to use on a personal VoIP setup. It is
strictly for my personal use, I'll never resell it, make money or it, or
whatever.
It seems a free personal-use G729 codec is available as a WIN32
library. I find it puzzling that at the same time one has to pay license
fees
>>>>> "Jan" == Jan Rychter <[EMAIL PROTECTED]> writes:
>>>>> "John" == John Todd <[EMAIL PROTECTED]> writes:
John> For what it's worth, I have noticed the same problem, but I think
John> the problem is in IAX2, since
> "John" == John Todd <[EMAIL PROTECTED]> writes:
John> For what it's worth, I have noticed the same problem, but I think
John> the problem is in IAX2, since my long-haul portions of the
John> diagram were over IAX2, while my SIP clients are almost always
John> sitting on the same LAN as th
[I have sent a message about SIP problems via gmane, but it seems the
list is gatewayed one-way only...]
The message was:
I've been trying to use Asterisk as a SIP->PSTN gateway. It runs fine
when the SIP client is on the local network and there is not packet
loss. But now I've tried running a r
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