I've problem plying * demo, sound breaks.
Connecting two h323 calls works fine, but when playing sound I get
message:
Request to schedule in the past?!?!
I've loaded ztdummy and usb-uhci but problem remains.
How can I tell that * is using ztdummy module?
lsmod says 'use by' for ztdummy is 0
Hi,
- Original Message -
From: Erick Weber V. [EMAIL PROTECTED]
Someone know wich is the best firmware for the ATA 186 with *
Version 2.16 (SIPH.323) works great for me (use it in production for more
than 6 months now without any problem).
BR,
Dan
Hi,
- Original Message -
From: Doug Harris
Where can I download this version ?
Cant find it here http://www.grandstream.com/TEMP/FIRMWARE/
Do you mean ATA-18x (from Cisco) or ATA-286 from Grandstream???
BR,
Dan
___
Asterisk-Users mailing
Did you create the various /dev/zap devices?
Does * automagicaly uses ztdummy or should I tell him to use it?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Nop. I'll make them.
Tnx.
On Thu, Mar 04, 2004 at 12:35:57AM -0800, [EMAIL PROTECTED] wrote:
Did you create the various /dev/zap devices?
Does * automagicaly uses ztdummy or should I tell him to use it?
___
Asterisk-Users mailing list
[EMAIL
Michael -
Here are out configs for our BT PRI which works:
I suspect it is the pridialplan entry which is missing. Also, you need
to have overlap sending enabled if you have overlap sending enabled on
your phones.
Another thing to look at is the actual digits being sent to line. in
your dial
We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently
it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's
what Cisco stated) but now we are hearing that it will not be fixed in that release
but would most likely be further down the
Hello, I have a X101P card and I contacted digium and they told me to return
it. Before I do so, I want to make sure the error is not on my side. If you
have any idea, please tell me. Here is my problem:
Recently setup asterisk and zaptel on a linux mandrake server and I got it
to work perfectly.
Now that i created devices ztdummy it is still unused module (lsmod)
How so?
Will ztdummy solve breaking sound when playing demo? (sched in past)
Dario
On Thu, Mar 04, 2004 at 12:35:57AM -0800, [EMAIL PROTECTED] wrote:
Did you create the various /dev/zap devices?
Does * automagicaly
Did somebody already get a snom 220 phone ??
DISCLAIMER: The content of this e-mail message does not constitute a commitment of
DKMA bvba This e-mail and any attachments thereto may contain information which is
confidential and/or protected by intellectual property rights and are intended for
Hi,
On my Suse90-out of box I had downloaded from CVS asterisk.
I'm running kernel 2.4.21-99-smp4g with 4cpu's, and the kernelsource is in
/usr/src/linux
Asterisk compiles with no problem.
But when compiling zaptel I got this error
..
zaptel.c: In function
Hi,
On Thu, 4 Mar 2004 at 11:29, Hans-Henrik Andresen wrote:
But when compiling zaptel I got this error
[...]
zaptel.c:5892: error: parse error before unsigned
make: *** [zaptel.o] Error 1
Any help on this ?
you left out the important part of the error log. I guess the compiler
complains
Greate -
/usr/src/linux/include/linux/version.h:6:2: #error The kernel sources in
/usr/src/linux are not yet configured.
/usr/src/linux/include/linux/version.h:7:2: #error Please run 'make
cloneconfig make dep' in /usr/src/linux/
/usr/src/linux/include/linux/version.h:8:2: #error to get a kernel
I use my HP ipaq with x-lite and it works
great
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
CollinsSent: Thursday, March 04, 2004 2:06 AMTo: Asterisk
UsersSubject: [Asterisk-Users] RE: Palm OS5
client
Does anyone know of a Palm OS5
client that can connect
I am currently running an asterisk server locally on a PC, to see what
kinds of things it can do. The output device is the sound system, via
alsa. It starts perfectly and plays the first sounds but after a minute
or to the sound dies and several messages
Mar 3 21:30:29 WARNING[180236]:
Hi,
Thank you for the information. There are ts in Dial command in
extensions.conf. When I deleted these ts, each sip phones were
directly communicating. I just wrote these ts from the examples.
Does these t and T are used for transfer(blind/consaltation) from
called user and calling user,
t and T are for # transfers. Other types of transfer are done in
other ways. Zap FLASH transfers are set in /etc/asterisk/zapata.conf.
I don't know how you enable/disable SIP or other types of transfers.
On Thu, 2004-03-04 at 06:51, Zen Kato wrote:
Hi,
Thank you for the information. There
One of the Possibilities could be that there is no disk space available,
or permissions. Do you get same error if you are root ?
I am currently running an asterisk server locally on a PC, to see what
kinds of things it can do. The output device is the sound system, via
alsa. It starts
It could be a case of the application trying to send more data than what
the sound driver can handle.
For debugging purposes, in chan_oss.c:181, you could try invoking
sleep() for a few milliseconds, without returning -1 right away. (of
course, for sleep(), the granularity is seconds, so you'd
Hello atif,
send an e-mail to [EMAIL PROTECTED]
I know nothing about voicemail and mysql configuration
--
Best regards,
Scottmailto:[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi all,
I'm experimenting with the following setup:
An Asterisk server at 192.168.0.10.
2 Linphones at 192.168.0.60 and 192.168.0.66.
The Linphones register themselves at the Asterisk as sip:[EMAIL PROTECTED]
and sip:[EMAIL PROTECTED]
If my understanding is correct they should be available on the
Hi...
Being very new to A* myself I understand your fustrations with the manuals
:)
It looks like you've made a typo in your extensions.conf
quote [sip]
extern = 66,1,Dial(SIP/66)
extern = 61,1,Dial(SIP/61)
extern = 60,1,Dial(SIP/60)
it should be
exten = 66,1,Dial(SIP/66)
Hope that helps
I'm partial to the v2.15 firmware, sip/h.323 and you can actually do a
factory default reset if you lose your password.
Thanks
-Matt
- Original Message -
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 2:29 AM
Subject: Re: [Asterisk-Users] Best ATA 186
OK, maybe I need more coffee. Or less. Either way, I'm stumped.
I have a Meetme conference room configured. Meetme(|M|) to enable
the MOH. When you are the first one to go into the conf, you get the
announcement that you are the only one, and then a *male* voice gives
a little talk about 'Why are
On Thu, Mar 04, 2004 at 02:53:20PM -, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Tim Sailer [EMAIL PROTECTED] wrote:
OK, maybe I need more coffee. Or less. Either way, I'm stumped.
I have a Meetme conference room configured. Meetme(|M|) to enable
the MOH. When you are the
I got bit by this today and was surprised to see the limit of a measly
100 messages hardcoded into voicemail. Is that right or am I missing
something?
Obviously, this should be moved to voicemail.conf. Does anyone know if
there's a reason why this hasn't been done, or if there's already a
I need to substitute a standard PBX. There are 70 intern analog phone lines, I mean, 70 analog extensions. Is this configuration possible?Is there a cheaper solution?
Asterisk box__ PRI | || TELCO |==OO|O|O|||= TE405P|___| |_||__ ||___||__ |
On Thu, Mar 04, 2004 at 02:06:52PM -, Jon Shamash wrote:
[snip]
it should be
exten = 66,1,Dial(SIP/66)
Incidentally, is there a difference between = and =, or are both allowed?
Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
simple answer: yes just use a good channel bank like an adtran or the adit600
On Thursday 04 of March 2004 16:17, Antonio Diego Almodóvar Cebrián wrote:
I need to substitute a standard PBX. There are 70 intern analog phone
lines, I mean, 70 analog extensions. Is this configuration possible? Is
Both are allowed but for readability = is used on objects.
Maxime
- Original Message -
From: Tor Houghton [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 10:19 AM
Subject: Re: [Asterisk-Users] 2 Linphones communicating through Asterisk?
On Thu, Mar 04, 2004 at
Silly me, I was
dreaming last night I guess. When I saw the post, I thoughtthat this is
Grandstream device.
BTW: Have you guys
seen new announcement from Grandstream.
http://www.grandstream.com/HT486.pdfat
85 MSRP.
DH
Message: 1
From: "Dan" [EMAIL PROTECTED]
To:
[EMAIL PROTECTED]
Hi guys,
I am kind of overwelmed with all the information in the asterisk site and
have no clue where to start. I have review some files but i am not certain
how to assemble all this. I got a dev kit with one fxo, and one fxs port. I
would like to setup my server to take incoming calls and hop
On Thu, 4 Mar 2004, FRANCISCO PEREZ-LANDAETA wrote:
Hi guys,
I am kind of overwelmed with all the information in the asterisk site and
have no clue where to start. I have review some files but i am not certain
how to assemble all this. I got a dev kit with one fxo, and one fxs port. I
ive been looking for a palm os5 client found gphone there webpage claims
to be sip but i just cant make it register against asterisk
Miguel
On Thu, 2004-03-04 at 07:05, Dean Collins wrote:
Does anyone know of a Palm OS5 client that can connect to asterisk?
Hopefully I can use gprs to connect
Hello,
I've got a small asterisk system running in production. We use the voicemail mainly
for letting people leave orders overnight and early morning in a special order
mailbox. The manager at the site keeps complaining that the messages are always split
into both old and new messages even
I've been told that MusicOnHold is *incredibly* picky about the mp3s
that it plays. I've experimented with the sample and a host of other
constant bitrate mp3s, and even some VBR ones, and I can't get any sort
of consistent workability. Even the sample doesn't work but maybe 10%
of the time.
Does anyone know of a way to reset the AbsoluteTimeout so it restarts
the timeout? i.e. regardless of how long the call is currently going
on, make sure it does not last more than 30 more seconds.
--Eric
--
Eric Wieling [EMAIL PROTECTED]
BTEL Consulting
Hi,
Recently I upgraded Asterisk from version 5 to 7
since I've done this all the calls that are private numbers are now showing up
as "Anonymous".
I know for a fact its not the Cisco 5300 striping
this off it appears to be Asterisk itself.
Does anyone know the section of source code that
I have a question about the capabilities on a user who wants to roam around and keep
the same extension. I see in the extensions.conf file you have to set the IP address
in there for the extension to call the phone. Is there a way to set the extension IP
to be setup per username instead. That
thanks, i will see if i can start...this.
- Original Message -
From: Andrew McRory [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 12:05 PM
Subject: Re: [Asterisk-Users] newbie
On Thu, 4 Mar 2004, FRANCISCO PEREZ-LANDAETA wrote:
Hi guys,
I am kind of
Jason Miller wrote:
I have a question about the capabilities on a user who wants to roam around and keep the same extension. I see in the extensions.conf file you have to set the IP address in there for the extension to call the phone. Is there a way to set the extension IP to be setup per
In article [EMAIL PROTECTED],
Daniel Prather [EMAIL PROTECTED] wrote:
I've been told that MusicOnHold is *incredibly* picky about the mp3s
that it plays. I've experimented with the sample and a host of other
constant bitrate mp3s, and even some VBR ones, and I can't get any sort
of
On Wednesday 03 March 2004 17:31, Serge wrote:
So, sorry I have general question , h.323 dont work on FreeBSD +
asterisk ???,,, I need converter h.323 sip and codec converter
for h.323. I use FreeBSD 5.2.
You've already answered your question. As chan_h323 does not
work on FreeBSD, and as
Yep, I have an x101p card that's up and operational. It has the wcfxo
and zaptel modules loaded. The error I usually get when trying to play
MOH is ...
Mar 4 12:29:43 WARNING[25618]: res_musiconhold.c:307 moh0_exec: Unable
to start music on hold (class '') on channel SIP/5000-4adf
It then
What processor? What distro? What kernel(SMP, non-SMP)? What are you doing
specifically with your Asterisk system?
My personal experience is that I've had high-load crashes anywhere from 6.0
to 8.0 on a SMP P4 Single-processor with HT enabled.
The load isn't the best indicator of when a crash is
On Thursday 04 March 2004 03:14, Maxime R wrote:
Hello, I have a X101P card and I contacted digium and they told me
to return it.
Digium is the final authority in these matters. If they believe that
you need to return the card, there is virtually nothing more that
anybody else can do to
Hi gang,
I've just set up my Asterisk server with a X100P (talking to a Vonage Motorola
do-dad), and a Cisco IP7960 SIP phone. All is working quite will with
outbound and inbound calling. However, I have a few questions.
First, regarding call waiting on Vonage/X100P, how do I click over to
On Thursday 04 March 2004 10:24, Jeff Roberts wrote:
I was just wondering exactly what all can happen that would mark a
message as old? Do messages left the previous day automatically
get moved to old?
Messages that have been listened to and are not explicitly saved to
the New folder are
On Thursday 04 March 2004 09:09, Matt Lawson wrote:
I got bit by this today and was surprised to see the limit of a
measly 100 messages hardcoded into voicemail. Is that right or am
I missing something?
It's related to a memory allocation during VoiceMailMain. If you
increase that number,
On Thursday 04 March 2004 11:16, PBXtech wrote:
At what point in the CPU load does asterisk start to fail?
CPU load does not measure what you think it does. It is simply
the measurement of the average number of processes in a short
wait status (i.e. waiting only for CPU time, not for disk
On Wed, 2004-03-03 at 16:12, Mark wrote:
The software configuration depends (of course) on your hardware
I have 2 Eicon Diva cards which I am using chan_capi.
I have chan_capi installed and configured and it detects the ports ok.
I have the lines plugged in but when I dial the number
Hello,
We have a * box up and running with a handful of SNOM 200 phones. It is
working very nicely.
I am trying to add an analog phone via a Cisco ATA-186 box. The ATA-186
registers fine. It will receive a call. But when it comes to dialing out
we get nowhere. :(
The ATA-186 is @
Hi Guys,
I anybody having problems with voicepulse out/in bound call ?
On the outbound calls im getting this error : (removed the username)
-- Executing Dial(SIP/103-296e,
IAX2/[EMAIL PROTECTED]/917707840009) in new stack
-- Called [EMAIL PROTECTED]/917707840009
Mar 4 12:51:31
Anyone in the New York City area do consulting on asterisk ?
Please contact me directly [EMAIL PROTECTED] thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Ah, so in a normal Asterisk world, the messages are supposed to be moved
to another dir.?
In our deviant Asterisk world, the voicemails are never checked through
the phone, only through a custom web interface, so they stay in INBOX
until they're deleted. Thus they collect quickly to over 100
Hello everyone,
Is there anyone know how to make fax detection work
with the T100P either to regular PSTN or VOIP? I'm having problem with
sending fax out. I have two T100P cards which connects to T1 PSTN and the
other connects to PBX (T100P -- Asterisk [T100P] --
PBX).
Thanks.
-Tri.
PBXtech wrote:
At what point in the CPU load does asterisk start to fail?
At the point you start getting blocked calls.
Jeremy McNamara
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
On Thursday 04 March 2004 10:24, Jeff Roberts wrote:
I was just wondering exactly what all can happen that would mark a
message as old? Do messages left the previous day automatically
get moved to old?
Messages that have been listened to and are not explicitly saved to
the New folder
Dear list,
does any one know how to do a SIP client auth via central database instead
of specifying in the sip.conf ?
if we could do with central database, should we use RADIUS or other better
way to do it.
Thanks,
George
___
Asterisk-Users mailing
Does anyone know if the 3Com NBX 2102 series phones with with * ? There
are a crapload (a very precise measurement) on eBay, but I can't figure
out what protocols they talk.
Tim
--
Tim Sailer Coastal Internet, Inc.
Network and Systems Operations PO Box
On Thu, Mar 04, 2004 at 11:24:12AM -0600, Tilghman Lesher wrote:
You've already answered your question. As chan_h323 does not
work on FreeBSD, and as you need chan_h323, you are therefore
required to not use FreeBSD.
Install Linux, like everybody else.
Genetic diversity in operating
Third, are there any VoIP providers that I can have Asterisk
talk to natively (i.e. via IAX or SIP, not the way I have
Vonage set up now)? I'd be looking for a Chicago land number
(630 specifically).
Check out www.iconnecthere.com - I think they've got pretty good coverage
nationally.
We're often getting a ringing noise or distorted voices during
conversations, especially when external callers are recording voicemail
messages. The system has only Digium FXO and FXS cards. (There is no IP
telephony yet.)
Can anyone help me fix this? Thanks.
Jim
On Thu, Mar 04, 2004 at 02:38:13PM -0500, Tim Sailer wrote:
Does anyone know if the 3Com NBX 2102 series phones with with * ? There
are a crapload (a very precise measurement) on eBay, but I can't figure
out what protocols they talk.
I believe they did make a version that spoke SIP, but
The 3com NBX phones use a protocol called H3, developed by graduate students
at MIT. The protocol has no documentation at all (except by a very P.O.ed
guy in Australia. 3com can by the protocol when they aquired NBX corp in
1999. (NBX corp was the company formed by MIT to market the system).
Hi Mark,
I'm having problem with fax detection on my Asterisk
box. My config is like this:
PSTN T1 == Asterisk (*) T100P (2 cards) ==
PBX
Everything works fine for voice and incoming fax but out
going fax got this error:
-- Starting simple switch on
'Zap/48-1'
-- Executing
Reece,
I have a similar setup by the sounds of things (running 0.7.2 with AS5300) and on
private number calls what you actually get is 'Anonymous 010101010101' and as far as
I remember it was always like that for me. How are you pulling the callerid into your
script ?
-Original
The original NBX100 phones spoke a proprietary voice-over-l2
ethernet protocol, but would upgrade to ip connectivity with
a liscense key on the NBX PBX box. There was an optional
software package that would let the NBX talk to an h323
gateway but it ran on nt and was rather klunky.
These
Well, I think I discovered even further why there is no ringback tone
available. The following message, is displayed on the console in asterisk.
ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device
Failed to register zone 'United States / North America': No data available
Looking more into
Hmm. Red button FACTRESET# works for me on 2.16 and 3.0
doesn't work for you?
Ejay Hire
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Matt
Sent: Thursday, March 04, 2004 8:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Best ATA 186
Just found on nwfusion.com:
3Com plans to announce the 3102 Business Phone, a SIP-based handset that
works with the vendor's VCX IP PBX, technology borrowed from 3Com's
now-defunct carrier softswitch business.
The phone is also compatible with 3Com's small- and midsize-site NBX IP PBX.
The
Hi,
I've had some issues with the x100p in my * box with echo at the beginning
of calls and remote hangup detection. Question is, will setting up an adtran
for the pots lines and connecting this to a t100p in * fix these issues ?
Can anyone verify good call quality and remote hangup detection on
On Thursday 04 March 2004 13:32, Jeff Roberts wrote:
On Thursday 04 March 2004 10:24, Jeff Roberts wrote:
I was just wondering exactly what all can happen that would
mark a message as old? Do messages left the previous day
automatically get moved to old?
Messages that have been
On Thu, 2004-03-04 at 13:48, William Waites wrote:
On Thu, Mar 04, 2004 at 11:24:12AM -0600, Tilghman Lesher wrote:
You've already answered your question. As chan_h323 does not
work on FreeBSD, and as you need chan_h323, you are therefore
required to not use FreeBSD.
Install Linux,
Hi. Asterisk doesn't currently support fax pass through as
far as I know. W/o fax pass through the faxes don't work
well at all.
-e
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of
Lach Dunlop
Sent: Thursday, March 04, 2004 11:49 AM
To: [EMAIL
They did make a SIP phone and are about to release new SIP phones and a new
product line. The old SIP phones look identical to the NBX phones but I am
not sure about the guts. Possibly the 2102 could be flashed into a 1002.
Here is an ebay auction but these phones are really hard to come by. I
Hello,
You have the ability to lock out the factory reset button, I feel more
comfortable just not having that option there.
Thanks
-Matt
- Original Message -
From: Ejay Hire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 2:38 PM
Subject: RE: [Asterisk-Users]
On Thursday 04 March 2004 13:48, William Waites wrote:
On Thu, Mar 04, 2004 at 11:24:12AM -0600, Tilghman Lesher wrote:
You've already answered your question. As chan_h323 does not
work on FreeBSD, and as you need chan_h323, you are therefore
required to not use FreeBSD.
Install Linux,
On Thu, 2004-03-04 at 14:47, Chris Clifton wrote:
Hi,
I've had some issues with the x100p in my * box with echo at the beginning
of calls and remote hangup detection. Question is, will setting up an adtran
for the pots lines and connecting this to a t100p in * fix these issues ?
Can
The 3com phones can't be flashed... they download their firmware image from
the NBX call processor when they power on...
However, if a SIP image you can have the phone download the image from a
linux box using the bootloader provided by Tim Hogard at
http://web.abnormal.com/~thogard/nbx100.shtml
I know a little history on the 3com SIP phones... We have about a dozen
of them
where I work. I'm not familiar with the NBX100 model number but the ones we
have are labeled: P/N: 655005001. The first ones didn't support SIP out
of the
box and had to be upgraded with a new flash image. I can't
Ariel Batista wrote:
Sorry I press send before I finished.
Chris Clifton wrote:
Hi,
I've had some issues with the x100p in my * box with echo at the
beginning of calls and remote hangup detection. Question is, will
setting up an adtran for the pots lines and connecting this to a
t100p in
Paul Vermette [EMAIL PROTECTED] wrote:
Well, I think I discovered even further why there is no ringback tone available. The
following message, is displayed on the console in asterisk.
ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device
Failed to register zone 'United States / North
On Thursday 04 March 2004 15:19, Ariel Batista wrote:
Chris Clifton wrote:
I've had some issues with the x100p in my * box with echo at the
beginning of calls and remote hangup detection. Question is, will
setting up an adtran for the pots lines and connecting this to a
t100p in * fix
Has anyone using the flash button on GS101 to access call waiting?
My experience is that it does not work. I read in the list that it may
need to tweak the flash duration to under 100msec. Has anyone have any
solution?
--
David Kwok
Tel: 612 99292086 ext 1002
Iaxtel/FWD # 17001813482 ext 1002
On Fri, 2004-03-05 at 02:04, dkwok wrote:
Has anyone using the flash button on GS101 to access call waiting?
My experience is that it does not work. I read in the list that it may
need to tweak the flash duration to under 100msec. Has anyone have any
solution?
GS, as in Grandstream
Tim Sailer wrote:
Just found on nwfusion.com:
3Com plans to announce the 3102 Business Phone, a SIP-based handset that
works with the vendor's VCX IP PBX, technology borrowed from 3Com's
now-defunct carrier softswitch business.
The phone is also compatible with 3Com's small- and midsize-site
Hello everyone,
If you don't have Digium card but you want to use
G.729 codec, do you need a license for it?
If the VoIP termination point supports G.729 and
you are using sip phone (soft/hard phone), can you use the G.729 pass thru or
you have to buy the license?
Have anyone test it with
Miguel
Can you let me know where I can find the gphone information so that I can
give it a try, thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Miguel
Cavazos
Sent: Thursday, March 04, 2004 5:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
Hi,
G.729 requires a license from Digium no matter
what. As long as asterisk doesn't have to encode anything for transmitting as
G.729 you can use passthru, this does mean however you will lose the asterisk
features such asvoicemail. The way thecodec for asterisk works is it
is based
As long as you have a IDE drive available, and mounted when you install it, it will
work. This includes CD ROM's...It's what I did.
Funkiness with the registration process.
As far as pass through goes, from what I understand, it *should*, but when you have
licensed binaries, from what I've
Thanks all,
I will install Linux. Your advice, what is better? RH 8 ? RH9 ? Mandrake?,
That was without problem.
I need converter SIPH.323 and H.323 codec converter.
Thanks.
Serge.
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March
On Thu, Mar 04, 2004 at 02:49:52PM -0600, Steven Critchfield wrote:
Genetic diversity in operating system support is a good
thing. It makes for more robust code. Following standards
is a good thing -- POSIX was written for a reason. If you
only support one OS you are less likely to
Has anyone encountered this issue:
I'm bringing up a T100P card and ran 'modprobe wct1xxp' but Zaptel Tool shows the card as UNCONFIGURED in its alarm window. the LED on the card itself is not lit and not flashing.
lsmod shows the module wct1xxp as zaptel both loaded.
Thanks in advance for
__
Hi,
I've had some issues with the x100p in my * box with echo at the beginning
of calls and remote hangup detection. Question is, will setting up an adtran
for the pots lines and connecting this to a t100p in * fix these issues ?
Can anyone verify good call quality and
On Thu, 2004-03-04 at 17:28, [EMAIL PROTECTED] wrote:
Has anyone encountered this issue:
I'm bringing up a T100P card and ran 'modprobe wct1xxp' but Zaptel
Tool shows the card as UNCONFIGURED in its alarm window. the LED on
the card itself is not lit and not flashing.
lsmod shows the
Thanks, from what I've gathered, remote disconnect supervision (hangup
detection) requires kewl start signaling. I currently have loop start
signaling ...will the telco change the signaling on the line ?
Or will the L36 firmware in the adtran make this work fine with loop start
lines ?
There
Miguel, I have since found out that my treo 600 (palm os5) cannot
utilise voice over gprs due to a problem with the microphone api (same
reason why it cant be used as a dictaphone).
I did however receive a number of comments from CE users (ipaqs mainly)
about how well they work. So don't give up
Darek,
Thank you for the info.
How is the sound quality when you are using with G.729 codec? What's your
thought?
Thanks.
- Original Message -
From: Derek Samford [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 2:58 PM
Subject: RE:
On Thu, 5 Feb 2004, Tim Sailer wrote:
Does anyone have the zaptel modules built for Debian 2.4.24 kernel?
After trying and trying to compile and make Asterisk run on a Debian
box, I gave up and picked another HD with RH 9 on it. No headaches. Only 1
build was necessary to build and run *.
I
1 - 100 of 105 matches
Mail list logo
