Jeremy McNamara wrote:
Mészáros Mihály wrote:
Please if you can please help me to solve this problem.
Help yourself and READ THE README.
Hello Jeremy!
I read it already! ;-) thx!
But i didn't find a word about that chan-h323 what decoder encoder use.
It use the libopenh323 or other in built encod
Except that £55 is more like $75-80 and not $35.
Regards, Wolf
"David J Carter" <[EMAIL PROTECTED]> writes:
I beleive Telappliant in the UK are doing them for £55, ($35)
http://www.voiptalk.org/products/index.php?cPath=27
Dave
Grandstreams are availabe for $65 quanity one, so its not hard to believ
Hello,
this is not really much of an issue any more in Europe, the old
state-owned monopoly phone companies have had to loosen up in the face of
private competition and de-regulation (or rather, fairly liberal re-regulation).
I something I hook up causes an actual technical malfunction in the
switc
There is no
USE_MYSQL_FRIENDS and USE_SIP_MYSQL_FRIENDS in .../asterisk/channels/Makefile
any more. But, on voip-info wiki it still says that it should be configured like
this. Anyone knows how should I tell Asterisk to use mysql database for SIP and
IAX friends?
Thanks
Tomica
Crnek
_
>From what I have seen so far on this list if you are running a version of
CVS-Head prior to release of Asterisk 1.0 then you should keep it and not
try to change or upgrade it. It would appear that there are a lot of recent
changes that may break if you try to upgrade to current CVS-Head, and
con
"dean collins" <[EMAIL PROTECTED]> writes:
> Lol, you're kidding right, go and look at what it costs to buy an
> alternative ip-pabx in comparison, and sorry but no corporate budget
> here, this is just a system for my home $100 on an old P3-700, and about
> the same on a card, and 2 $55 grandstrea
I have been running Asterisk happily for many months and I was trying to
upgrade from CVS-HEAD-08/13/04-10
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of el Flynn
Sent: Monday, 11 October 2004 16:09
To: Asterisk Users Mailing List - Non-Comm
Simon Brown wrote:
I have just downloaded V1.0 from CVS and when I try to start Asterisk (after
compiling and installing) I get this error:
Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:6205 mkintf: Unable to get
parameters
Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:9134 setup_zap: Unable to
reg
On Mon, 11 Oct 2004 00:10:26 +0100, David J Carter
<[EMAIL PROTECTED]> wrote:
> I beleive Telappliant in the UK are doing them for £55, ($35)
>
> http://www.voiptalk.org/products/index.php?cPath=27
>
> Dave
£55 is more like US$100 :-)
___
Asterisk-User
I have just downloaded V1.0 from CVS and when I try to start Asterisk (after
compiling and installing) I get this error:
Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:6205 mkintf: Unable to get
parameters
Oct 11 15:51:29 ERROR[1076241024]: chan_zap.c:9134 setup_zap: Unable to
register channel '2'
Hi again, folks
A short update
On Sun, Oct 10, 2004 at 12:41:07PM +0200, Tzafrir Cohen wrote:
>
> We have created a simple Debian-based distribution of Asterisk. A CD
> image of an installer(150MB, requires no extra packages from the 'net)
> that installs Debian and Asterisk simple and easy.
>
Has anyone
experienced this problem? The C flag in Dial app doesn't work. I'm getting the
CDR record although it has C (reset CDR for this call). The C is even recorded
in the CDR record lastdata field.
Assaf
BenharooshMCP, MCSA, MCSE[EMAIL PROTECTED]
__
I want to build a conferencing system and I'm looking for suggestions on
which application will make a better starting point.
A conference consists of: up to 20 callers, a small number (1-3) of agents
and a small number of supervisors (0-3). Multiple conferences will be
"active" on the same hos
On Mon, 11 Oct 2004 00:10:26 +0100, David J Carter
<[EMAIL PROTECTED]> wrote:
> I beleive Telappliant in the UK are doing them for £55, ($35)
Whoa, that's an amazing exchange rate you've got there. I'm sure at
that rate some American cowboy will buy another London bridge and New
York taxis are goi
On Mon, 2004-10-11 at 00:10 +0100, David J Carter wrote:
> I beleive Telappliant in the UK are doing them for £55, ($35)
>
> http://www.voiptalk.org/products/index.php?cPath=27
Your conversion above is going the wrong way. a British pound is worth
more than a US Dollar. In fact, 55 British pounds
On Sun, 2004-10-10 at 21:34 +0100, Kevin Walsh wrote:
> Brian Capouch [EMAIL PROTECTED] wrote:
> > Wolf Paul wrote:
> > > If that warrants "don't come asking for support" then you guys are not
> > > much of a community but a sales
> > > machine for Digium.
> > >
> > Digium does Asterisk FOR FREE.
try it on the same lan.
- Original Message -
From: "Sudhir Kumar" <[EMAIL PROTECTED]>
To: "Steve Totaro" <[EMAIL PROTECTED]>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Sunday, October 10, 2004 7:50 PM
Subject: Re: [Asterisk-Users] SIP device n
Because realtime isn't in 1.0 or 1.0.1 its ONLY in cvs-head.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of mihai iancu
> Sent: Sunday, October 10, 2004 9:05 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] newbie question
Simon wrote:
i have already configured a sipura 1000 , xlite client and analog phone
but i wanted to connect a fax line to the sipura 1000 so when i
receieved a call and detected to be a incoming fax to be diverted to
sipura 1000
please guide what to change and where to change
thank you so much
Hello,
Here are my info: asterisk version 1.0 with Redhat 8.0 kernel 2.4.18
Everything was running nice and clean with an old version from Aug
2004.
Cleaned all source code and binaries - download and install version 1.0
and this is what I get:
Oct 10 22:44:36 WARNING[8192]:
/usr/lib/asterisk/m
i have already configured a sipura 1000 , xlite client and analog phone
but i wanted to connect a fax line to the sipura 1000 so when i
receieved a call and detected to be a incoming fax to be diverted to
sipura 1000
please guide what to change and where to change
thank you so much
Simon
___
I am running the latest asterisk CVS.
[EMAIL PROTECTED] /etc/asterisk # locate demo-thanks
/var/lib/asterisk/sounds/demo-thanks.gsm
This directory has 150+ files.
I changed the [demo] section in extensions to [incoming] to play with
this included definition:
[incoming]
;
; We start with what to d
might want to check you port card and be sure you have fsx
ports rather than fxo ports...
I'm running a II with stock configs and having no
problems.
tim mckee
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mason
HerringSent: Sunday, October 10, 2004 3:14 PMTo: 'Aste
Technically speaking you would install it directly on the relay coil leads.
It may seem like semantics, but if you had a cutout switch or other
electronics between the power supply and the relay, the diode needs to be as
close to the relay as possible electrically.
BTW, I have only send a few thou
OK... so what you're saying is that I put a diode across the power supply input legs for the DPDT
relay, right?
(sorry, i'm not the best person at electronics...)
Greg Hill wrote:
On Sun, 10 Oct 2004, Rajeev Sharma wrote:
Yeah, thanks, I was thinking of doing something similar to that.
Actually,
Hmm, something must've gone wrong with the quoting below I didn't
write that which was attributed to me here!
Greg
On Sun, 10 Oct 2004, Jerry Glomph Black wrote:
> Not seen unlimited flatrate? You are not looking very hard.
>
> Umm, you are ignoring Broadvoice, which has no crappy bonding t
On Sun, 10 Oct 2004, Rajeev Sharma wrote:
> Yeah, thanks, I was thinking of doing something similar to that.
> Actually, I was gonna spice a cable in my computer's power supply and
> use that. Why? Because if it's on a UPS, then the switch will throw at
> the same time as the computer looses off.
Hi,
We bought one of these gateways 6 months ago, it was shipped to us
3 months after our order was placed and damaged enroute due to inadequate
packaging.
It arrived with no software or manuals and nobody at Flosys returns calls
or follows up on emails or voicemails.
Does anybody else here us
Hi all:
I've setup Asterisk as a SIP server running on port 5060. I've also
added port=5061 and host=127.0.0.1 to user 'me'. As a result, I can
use Linphone and Asterisk on the same machine. Everything works except
the microphone.
I'm using ALSA drivers for VIA-823* on kernel 2.6.8.1
I noticed t
Could these files be cached as well?
Donny
-Original Message-
From: Paul Dugas [mailto:[EMAIL PROTECTED]
Sent: October 10, 2004 12:56 PM
To: Asterisk Mailing List
Subject: [Asterisk-Users] TTS via text2wave
Tinkering with getting a text-to-speech component worked into my dial
plan with
If you are going to buy it new. It should not be a problem at all. Just
order the SIP software with the phone.
Your order should be somthing like this.
CP-7905G - Cisco IP Phone 7905G, Global
SW-SMH-UL-7905 - SIP or H.323 license for single 7905 IP phone
CON-SNT-CP7905 - 8x5xNBD Svc, Cisco IP P
Be careful Cisco EOL'd this product, the 7905 IP phone, in 2003 and as of
August 8th 2004 does not have to support any software. The actual last day
of support that you will be able to download software even if you have a
contract is August 2005. The 7905G is another story, this replaced the
7905.
I had connected to asterisk with "asterisk -vvvr" but did not see
anything about registration (accepted or denied) from that device. Thats
why I suspected perhaps ISP is blocking something.
Thanks,
-- sudhir
On Sun, 2004-10-10 at 13:15, Steve Totaro wrote:
> some output from asterisk console woul
This line was reprovisioned, and the password was changed.
Stan
Stan Brinkerhoff wrote:
Hello,
I have been trying to setup * with Broadvoice.
I am using Gentoo Linux, and * 1.0.0, and now CVS.
My current config looks like:
(sip.conf)
[general]
port=5060
context=sip_incoming
tos=lowdelay
notifymim
Hello,
I have been trying to setup * with Broadvoice.
I am using Gentoo Linux, and * 1.0.0, and now CVS.
My current config looks like:
(sip.conf)
[general]
port=5060
context=sip_incoming
tos=lowdelay
notifymimetype=text/plain
allow=gsm
allow=ulaw
allow=alaw
canreinvite=no
nat=no
register => 8027051
> i need to know the correct procedure; otherwise i will bringing my
> customers in danger and that is not what i want.
>
> i know you can buy the 7905 WITHOUT the callmanager license.. if i
> load the sip image in it; will that be ok ?
Its my understanding (which could be somewhat incorrect) th
I beleive Telappliant in the UK are doing them for £55, ($35)
http://www.voiptalk.org/products/index.php?cPath=27
Dave
Grandstreams are availabe for $65 quanity one, so its not hard to believe
that you could get them
for $55 for larger quantities
http://froogle.google.com/froogle?q=grandstream
I'm the same way... Asterisk, zaptel and libpri are all done from src and
not portage.
bkw
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Karl Dyson
> Sent: Sunday, October 10, 2004 3:54 PM
> To: Asterisk Users Mailing List - Non-Co
You can't get the SIP image for the phone without a support contract with
cisco. Search the list, this is well documented.
Matthew
- Original Message -
From: "Danny Zak" <[EMAIL PROTECTED]>
To: "Kannaiyan Natesan" <[EMAIL PROTECTED]>
Cc: "Asterisk Users Mailing List - Non-Commercial Discu
Hello Kannaiyan,
i need to know the correct procedure; otherwise i will bringing my
customers in danger and that is not what i want.
i know you can buy the 7905 WITHOUT the callmanager license.. if i
load the sip image in it; will that be ok ?
--
Best regards,
Danny
Mészáros Mihály wrote:
Please if you can please help me to solve this problem.
Help yourself and READ THE README.
Jeremy McNamara
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
oo. Is there is anything such so.
When you buy it for you, does that device not belonged to you?
I don't think cisco can catch you if you load linux on that and use that as
a computer.
-Kannaiyan
- Original Message -
From: "Danny Zak" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing L
Hello Asterisk,
can somebody tell me what to purchase EXACTLY for not violating the
licensee agreements of this phone in according to use it properly
with *
--
Best regards,
Danny mailto:[EMAIL PROTECTED]
belGOnet.com a Euro-pictures division - internet soluti
Grandstreams are availabe for $65 quanity one, so its not hard to believe that you
could get them
for $55 for larger quantities
http://froogle.google.com/froogle?q=grandstream&hl=en&lr=&tab=wf&scoring=p
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: "Steve Tot
Personally I get zaptel, zapata etc from cvs rather than portage.
Check your /usr/src/linux symlink points to the correct place... I got
all sorts of grief with gentoo when I forgot to put it back after some
"playing".
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTE
OK, I got a little further. I don't know why but after re-emerging zapata
finally zaptel will build. It is not working however and I still get too
many erros during the build. It's now complaining about:
CC [M] /var/tmp/portage/zaptel-1.0.0/work/zaptel-1.0.0/zaptel.o
/var/tmp/portage/zaptel-1
Brian Capouch [EMAIL PROTECTED] wrote:
> Wolf Paul wrote:
> > If that warrants "don't come asking for support" then you guys are not
> > much of a community but a sales
> > machine for Digium.
> >
> Digium does Asterisk FOR FREE.
>
I wish people would stop posting misleading comments like that.
A
Brian West [EMAIL PROTECTED] wrote:
> Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
> >
> > Having said that, you have a good case in favour of the Intel modems
> > if you are in a country where the X100P doesn't have type approval but
> > you can find an Intel modem (with the right c
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