:
Andres wrote:
Ing. Angel Gomez wrote:
Hi, i just received an E100P, this is the first one I have ever seen,
and notice that the board reads T100P. Is this right ?
I think this was asked just a few days ago...the answer is YES.
If people would read the included documentation from Digium
Hi, i just received an E100P, this is the first one I have ever seen,
and notice that the board reads T100P. Is this right ? The antistatic
bag had a small label that has E100P written on it, and the card is a
bit different than the T100P I already have, Does Digium use the same
boards for
Connect them via IAX2 and use the switch instruction on the
extensions.conf of PBX200.
Deepak Malhotra wrote:
Hello
I am trying to setup Asterisk on 2 servers PBX300 and PBX200.
PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap
device.
Softphone from PBX200 can talk to
Hello all.
I saw a few weeks ago a discussion about cal pickup, *8, not working
but did not find a message about it being resolved, I look for a bug on
the bug list but did not find anything about it not working, nor a bug open.
I installed asterisk 0.9.0, have one sip fxo gateway and
Hello.
I have been going thru the wiki and asterisk related sites and have not
been able to find any documentation about how to configure an Adtran
TA750 channel bank.
The remote disconnect supervision doesn't seem to be working, when the
remote caller hangs up asterisk takes up to 30-45
Walter Doerr wrote:
On Mon, Jan 19, 2004 at 12:33:41PM +0100, Philipp von Klitzing wrote:
Ok,
here comes part two of the log quiz, this time SIP not MGCP:
WARNING[8201]: chan_sip.c:4821 handle_response: Got 200 OK on REGISTER
that isn't a register
This is most probably cause by
Hi all.
I have this configuration:
Telco -(E1)-TE410P//Dual Xeon Server
2.4Ghz-(Ethernet)-Switch-GS//BT
The Server is running RedHat Linux 8.0 with kernel 2.4.18-14-smp and
we are having the following 2 issues:
1.- When making calls from the GrandStream to the
Hi all.
Could it be possible that video frame buffering be causing problems
even if the computer is not running X ?
Thanks.
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James Sharp wrote:
Hi all.
Could it be possible that video frame buffering be causing problems
even if the computer is not running X ?
Yes. There are known problems with systems running with either a frame
buffer console or a serial console. For best results, run a plain VGA
Hello all.
What is ztcfg for ?, what does it do ?
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Hello.
Do you have your linux starting in Graphical mode ( init mode 5 ) ? I
also had a problem with audio on my sip phones and it was generated
because of the frame buffering that my video drivers use ( I have *
installed in my personal computer ), so I changed the startup mode to 3
and only
to it if it is too busy. good sign isnt it
Go get a cup of coffee, try it again you should be able to get it..
Ing. Angel Gomez wrote:
Hi all.
Is there a problem with digium cvs ? I can't connect to it, it just
keeps giving a...
cvs [login aborted]: connect to cvs.digium.com(216.207.245.20
Done.
Thank's all.
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Steve Bradwell wrote:
Hi All,
Just installed our very first asterisk system, and we love it! I cant
believe the different things you can do with it, just great =]
My question is: How do I configure my system to play an mp3 file when a
caller gets put on hold?
Thanks in advance,
Steve.
Hi all.
Is there a problem with digium cvs ? I can't connect to it, it just
keeps giving a...
cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401
failed: Coonection refused
Thank's
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Good afternoon.
I got my hands on a couple of zplex channel banks, they are
suppoused to work with asterisk, and have a few questions ...
1.- Where can I get a recent firmware for it ? The swversion shows
zplex10b rel 1.0.1.
2.- In the back of the channel bank says, model:
WipeOut wrote:
Jason A. Pattie wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
Good evening all.
I'm having this strange behavior when dialing two or more
simultaneus calls via IAX to other * boxes. Sound starts to have more
latency, wich increments until it's almost impossible to talk (6 or more
seconds), I try this calling with two grandstreams, one grandstream
Ing. Angel Gomez Garcia wrote:
Good evening all.
I'm having this strange behavior when dialing two or more
simultaneus calls via IAX to other * boxes. Sound starts to have more
latency, wich increments until it's almost impossible to talk (6 or
more seconds), I try this calling with two
Bisker, Scott (7805) wrote:
Just submitted a patch for this on asterisk-dev
GGrreeaatt!!
Will test ASAP.
Thank's.
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Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
Thank's.
WipeOut wrote:
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen
Bartosz Jozwiak wrote:
Hello,
How to turn off native bridge in Asterisk.
Is it possible ??
Bart
canreinvite = no
in each entry your sip.conf
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Hi.
Do you have mpg123 installed and mpg123 in /usr/bin ? I have moh
working with snom200 and there was no issue to have them working.
I even put an extension in my extensions.conf so the user can dial
it an hear the music, cause the snom200 has call waiting they don't miss
calls
Did you modprobe ztdummy before running asterisk ? I have meetme
running in one * box without zaptel harware.
rnc Info Lists wrote:
Yes, I am a newbie too. I am having a problem with meetme. From what I
have seen it will work without a Digium card but with audio problems. My
goal is just to
Thank's.
Ing. Angel Gomez Garcia wrote:
Hello.
I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and
i dont get a ring in the caller phone when I dial from a snom200 to
the other snom200 or the mp108fxs, I made a debug with ethereal, and I
can see a Ringing packet being
Steve Lorimer wrote:
I've setup a trial Asterisk install based on the RH8 install guide
mentioned on this list. (Thanks, Andy!) I've configured two other
working systems with SJPhone software for SIP. However, while I can
call the phones, the communication is choppy. About every three or
Hi.
Do you know what switch your telco has ? The one they are using to
provide you the service.
Osvaldo Mundim Junior wrote:
Hey all!
I had an experience trying to set up an E1 in Brazil which could help
somebody. In Brazil is very common telcos to have just R2 digital as their
primary
Hi.
I am not using cache, just :
festival.conf
-
[general]
host=localhost
port=1314
festivalcommand=(tts_textasterisk %s 'file)(quit)\n
but in extensions.conf when i call the festival app i put the text
'quoted' like this:
exten = 003,1,Festival('Hello asterisk
Hi.
I have the next configuration... I dial from my analog phone in the
TDM400P to extension 102, and the second agi works about 1 out of 10
times, the other nine it gives me these error on the asterisk console:
-- Starting simple switch on 'Zap/2-1'
-- Executing Macro(Zap/2-1,
Hello.
Does someone have a patch for SayNumber function (say.c) for numbers
in spanish ?
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Hello.
Has someone been able to make work faxes over sip, i have one mp108
fxo and one mp108 fxs, my setup is :
telco analog line - mp108fxo - Asterisk -- mp108fxs
--- fax machine
1) Asterisk detects the tone from the sending fax ( i am receiving ) but
looks for
WipeOut . wrote:
now .. i have one slight problem left .. although most of my SIP
phones are on a LAN connection with the asterisk server,
there are two phones which are at a remote office bridged to
my LAN via a 128k point to point ADSL .. these do not seem
to be working well, you do hear speech
Ajit Kallingal wrote:
Hello
After having installed my X100P, /proc/pci and /proc/interrupts dosent
locate them.
In my PCI list is it the Communication controller: Tiger Jet Network Inc ?
then is dosent have a IRQ listed in /proc/pci..
All help appreciated
Thanks and Regards
Ajit
it configured for US and Mexico
Thanks
Tan
telappliant.com
- Original Message -
From: Ing. Angel Gomez Garcia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 21, 2003 6:39 AM
Subject: Re: [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway
(SIP)
Hello Ernest.
I'm setting
Try ftp://angelgomez.homelinux.com/pub/audiocodes
Anton Tinchev wrote:
Kelvin Chua wrote:
hi guys,
have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing?
~kelvin
Can someone send me SIP firmwire for audiocodes 104.
I has h.323 only and it sucks
Kelvin Chua wrote:
hi guys,
have anybody tried using audiocodes sip fxs against asterisk? how's
the device fairing?
~kelvin
Yes, Ok.
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I am using the audiocodes MP-108 FXO as a SIP gateway to the pstn,
it works fine, and asterisk can receive and make calls thru it, caller
id is also working ok. As for now the only remaining issue that i have
pending is with conferences, the channels of the lines that get into a
conference
Hi all.
Just got my 'Developer Kit Lite', installed it, and made the changes
to load the modules in kernel and in the configuration files. Call
thru-from fxo and the fxs sound great. Even fxs-iax-sip sound ok.
When a answer a call coming into asterisk from the PSTN thru the fxo
i
Hi all.
Is there a way to collect the digits dialled in asterisk and stored
them in a variable ? I'm setting a submenu for the user to change his
extension dial in treatment from a standard extension to something like
'automatic transfer' and I need to ask for the number where to
could match a number and store it in a db.
see http://www.junghanns.net/asterisk/page6.html
(the great capigod asterisk page)
Matteo.
Il sab, 2003-07-05 alle 12:00, Ing. Angel Gomez Garcia ha scritto:
Hi all.
Is there a way to collect the digits dialled in asterisk and stored
them
Yes, look into the file cdr_mysql under asterisk cvs directory (
/usr/src/asterisk/configs/cdr_mysql.conf.sample )
God Knows Well wrote:
Hi
I think * supports database integration i would be thankful if anyone
help me to configure my asterisk box with database support. One more
think can i
I already have a web page to do it, I just wanted to add these feature
upon request by a customer, and yes, he wants the data keyed in.
I was looking at AGI command, GET DATA, will try it.
Thank's
Steven Critchfield wrote:
On Sat, 2003-07-05 at 05:43, Ing. Angel Gomez Garcia wrote
I had a similar problem and solved it changing the params of input
gain on my pstn-gateway, change from a value of 10 to a value of 1 and
that eliminated the echo on the SIP Phones.
Dave Packham wrote:
Same prob here. 15 SIP phones only get eco when going to the PSTN...
if you find
in zapata.conf file?
It is about rxgain?
BR,
Dan
- Original Message -
From: Ing. Angel Gomez Garcia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 11:48 AM
Subject: Re: [Asterisk-Users] Problem with echo
I had a similar problem and solved it changing the params of input
Yes you can. Configure it either as a SIP gateway or an h.323
gatekeeper.
Bradley Greep wrote:
I've been reading the Linejack strikes again messages, and have another Newbie question
is it possible to use a Voip Product as a Dialout line for * ?
I have a Vegastream 100 Voip to PRI. box.
AudioCodes has one 24 port fxs sip interface, i have one 8 port fxs with
SIP up and running with *
Michael Kane wrote:
The Cisco 242x (20 or 21), has a 24 port analog interface that supports 16
FXS and 8 FXO. I've delpoyed hundreds of these IAD's signaling with MGCP.
Not sure if it supports
hostname=localhost
dbname=asteriskcdrdb
password=
user=asteriskcdruser
carlos del mayor wrote:
I'm only asking for some examples of cdr_mysql.conf, nobody has
done anything with cdr and mysql? If you think is better another DB,,,
tell me, please!
thanks in advance
carlos
carlos
You must have a Zaptel device installed in your computer or load
ztdummy module to get conferencing to work...
Jordan Peterson wrote:
I don't know what that is, so probably not. Is that a conference type
board? Is there a way to make conferencing work or to assign an
extension to a h323
denon wrote:
At 06:44 PM 6/11/2003 -0400, you wrote:
On Wed, Jun 11, 2003 at 12:42:57AM -0500, denon wrote:
We're doing a new * installation at a remote office soon, and I was
just
curious what people's opinions were on hardware these days .. I've had
decent luck with T100Ps and Adtran, but
Good evening.
I have a problem with my Xpressa phone, when i dialed from/to it i don't
get audio, my other UA are a SJPhone and XLite, i already debug it with
ethereal and tcpdump, i dialed the echo test extension from the demo
files of asterisk and is the same result, no audio/rtp coming from
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