Hi armin,
thx for the answer. I have connected the BRI on a
HiPazt4000 and i still have the same issue. So i think
i have a problem with my ISDN line. I will contact my
provider. May be a reset of the line will solve the
problem.
rich.
--- Armin Schindler [EMAIL PROTECTED] wrote:
On Tue, 17
Yeah, Paul. I guess you're right..
Just tested speex and got complains from my customer :S..Maybe this
codec is not suited for our network ;)..
Regards,
Stevanus
[EMAIL PROTECTED] wrote:
Quick question - what is the point of speex? Do we really need it as an
option?
PaulH
-
Hi
Srs.,
we have installing two machines with Asterisk and Linux-HA. I just copy
conf files and voicemail files and more with rsync, and now I want to test with
Linux-HA if asterisk is up. I'm installing Asterisk over Debian, but I haven't a
status function in asterisk script.
Any one
hi,
it seems that spandsp-0.0.2pre22 is not functioning right. downgrading
to pre21 makes it work again.
debug messages:
Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Changed from phase 1 to 4
Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW ???:
Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW
On Wed, Jan 18, 2006 at 03:44:03PM -0800, calvis wrote:
Sorry about the OT thread, but I am sure that someone could give me some
advice. Nothing is more frustrated than doing a long cable run and then
finding your cable is defective.
OK, I have had it with the General Cable brand of
On Thu, Jan 19, 2006 at 03:46:41PM +1100, [EMAIL PROTECTED] wrote:
Quick question - what is the point of speex? Do we really need it as an
option?
Three points:
1. some nice features, see http://speex.org/ . Some are unique. Though
not all are applicable to transcoded usage with Astrisk.
2.
On Thu, 19 Jan 2006 00:16:19 -0600, Russ Price wrote
Ross C wrote:
We've always used Coleman and Belden with great results. We get a good deal
on Coleman, so that's what we usually use--never had a problem. Berk-Tek is
also good, I know a lot of cable installers who swear by it (I think
On Mon, Jan 16, 2006 at 08:17:28PM -0600, Moises Silva wrote:
I dont know the wakeup feature. But what you want can be done with a
web interface generating .call files with the timestamp of the day,
hour and time when you want to hear the reminder. Just read in
voip-info about the .call files
I have seen the following effect in Asterisk, though: where
it converts
an inband DTMF (eg coming off a Zap channel) into an
indication, it mutes
the audio where that tone is. But sometimes it leaves a
teeny bit of the
tone behind.
If you take such a call over say IAX to
Where can I investigate the origin of brief silences during calls from/to my
SIP phone?
Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything.
Thanks
Mimmus
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
Dear All,
I am having a problem in a scenario I am doing, I have two branches,
every branch has has an [EMAIL PROTECTED] that deals with each branch locally
and a trunk connected to a central asterisk, now if any branch wants to
call another branch it goes from the local asterisk@ home --> to
-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160
[EMAIL PROTECTED] wrote:
| On Wed, 18 Jan 2006, Hirosh Dabui wrote:
|
| look there http://snom.com/wiki/index.php/Xmlobjects for snom
| 360...
|
|
| nice... any hope for snom 320?
|
| -Dan ___
|
i
Under the regional tab (admin/advanced) select
ETSI FSK WITH PR (UK) for caller id method.
Make sure
PSTN CID For VoIP CID: is set to yes in the pstn tab to pass on the cid to
asterisk.
Chris
- Original Message -
From: Conrad Wood [EMAIL PROTECTED]
To: Asterisk Users Mailing
You might try using a tool like Ethereal to look at what's happening on the
network.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mimmus
Sent: 19 January 2006 10:13
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
I
have aproblem with the cdr.
We terminate through
a pstn provider to the pstn network.
The problem is now
the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn
number.
So i have billsecs
all the time even it is only ringing or so.
It seems that ETSI standard ES 201 912 documents the protocol which is
(may be?) used in Australia.
If anyone is interested it can be downloaded from
http://www.etsi.org/services_products/freestandard/home.htm, after
filling in some soul sucking registration details :)
James
-Original
[EMAIL PROTECTED] wrote:
On Wed, 18 Jan 2006, Javier Oviedo wrote:
Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor:
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11
seconds
Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback:
Pisac wrote:
I'm very surprised that Asterisk (PBX !) do not support AOC.
Setting some variable with AOC informations should be enough.
Storing AOC in CDR would be perfect.
Arg!
I noticed just now that Asterisk breaks all my report applications,
extracting accounting data from an
Hi,
On Wed, 18 Jan 2006 21:18:47 -0200 Juan Carlos Castro y Castro [EMAIL
PROTECTED] wrote:
| OK, I've got the new schema installed and I was able to create
| oxyPBXAccountSIP objects.
|
| Now, how do I generate the MD5 values to put in the realmedPassword field?
md5sum -b
Hi,
On Wed, 18 Jan 2006 18:50:33 +0530 Chandan Mishra [EMAIL PROTECTED] wrote:
| Hi
| I need to authenticate all the asterisk users from the LDAP server instead
| of from sip.conf.
| If anybody already have done this then please guide.
|
| I tried to integrate authenticate asterisk users
René Enskat [Teamware GmbH] a écrit :
I have a problem with the cdr.
We terminate through a pstn provider to the pstn network.
The problem is now the cdr accounts the connection to the gateway. Coz
the gateway is answering our call and then forward to the pstn number.
The gateway is
Have you seen if this equipment share IRQ for the resto of PCI Slots. I want
to install one TDM2400P with 24 FXS Port and one TDM04B with 4 FXO ports but
I want to know if that equipment has voltage connector for TDM2400P and it
doesn't share IRQ in two PCI Slots.
regards,
tron
-Mensaje
I am now getting these messages on a second box running a different version
of Asterisk. If anyone has any idea what is causing these, or how to avoid
them I would be very grateful.
157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on
Local/[EMAIL PROTECTED],2 of format ulaw
Hi All,
I tried with different configurations and referred many articles to configure
Asterisk with a Vonage account I have but all my attempts failed. I am a newbie
and hope this mailing list will help fixing my problem and configure Asterisk.
The error I get after I make a call to outside
Hello,
I did some tries with efax and it worked pretty well, however I sent
faxes to the same machine, so I don't know how good is this combination
in real life faxes.
Regards,
Tamas
Carlos Chavez wrote:
Has anyone tried to use an Asterisk server with iaxmodem and
efax? I have
Hello Everone,I hope everyone is having a good day. I am having a problem with my asterisk box.
When I call the box from a land line or cell phone and I press a number
I hear a very loud tone and then it comes back and says the person is
unavailable.
The loud tone I hear is very annoying. I
I wonder whether anyone got the Sipura ata 3000 to decode British
Telecoms callerid and pass it to asterisk?
The userguide seems to suggest that this is not possible, is that right?
It's definately possible... A quick google returned the following (from
Can someone give me an idea
of the processor power I will need for 1 x TDM240 with 2xquad FXO's and
8 sip phones/ATA's on a quite 100Mbit LAN.
The machine we have available of hand is a P4 1GHz with 768MB RAM.
Tx
Marnus van Niekerk
--
"Opportunity is missed by most people because it is
I wrote a connection pooling patch because asterisk is not usable with
MSSQL without it.
If you're using, or would like to use, MSSQL I recommend you to check it
out.
http://bugs.digium.com/file_download.php?file_id=8809type=bug
Just so you know, this is a diff against 1.2.1 and it's been only
Marnus,
Sounds like a good configuration
to me.
You might want to upgrade
the RAMs to over a gig, but thats about it.
Joash
www.kahuna.nl
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk
Sent: Thursday, January 19, 2006
2:23 PM
To:
Marnus - I assume you mean a Pentium "3" 1GHz
processor? I don't believe there was a 1GHz P4. You should be
alright with that machine, you can always add more RAM, RAM is pretty
inexpensive these days.
Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY
Thatshouldbefine.Ihaverunmorewithmuchless.
As a rule disable any uneeded services (chkconfig is
your friend).
Also it has been mentioned time and time again DO NOT
RUN X on the machine. The load placed on the machine will interupt your Asterisk
process and you will get choppy sound at
I have installed the sipTAPI from http://sourceforge.net/projects/siptapi/
when I use user names like joash.herbrink in
Asterisk, it is not working
when I change the sip username to my internal
extension, like 1006, it works fine.
Anybody any idea as to why this is?
met
Thanx for all the answers. I
do intend using chkconfig to disable everything and definately no X.
This machine will be asterisk only - nothing else. (Except httpd and
mysql for web based cdr reports.)
M
Alexander Lopez wrote:
Thatshouldbefine.Ihaverunmorewithmuchless.
As a
On Wed, 2006-01-18 at 14:26 -0500, Adam Robins wrote:
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
Have a look at www.dothill.com. SANnet II boxes
[EMAIL PROTECTED] wrote:
I have seen the following effect in Asterisk, though: where it converts
an inband DTMF (eg coming off a Zap channel) into an indication, it mutes
the audio where that tone is. But sometimes it leaves a teeny bit of the
tone behind.
Yes, that is correct. By the
yes, asterisk work with centos
- Original Message -
From:
Eric
Bishop
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, January 19, 2006 1:47
AM
Subject: Re: [Asterisk-Users] asterisk
1.2.2 RPMS for CentOS 4.x
will they work with
i'm finding a little script example in phpagi, to do a query in mysql, how i
do that, beacause i'm tired of finding information about that,
and the code of php dont work for me
anybody have a little example on how do that???
__
Visita http://www.tutopia.com y
We need an expert in least cost routing (LCR) for an Asterisk project.
Please provide references and a resume of your experience. Contact us at
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
Here is a simple mysql snippet in php. Straight from the PHP manual.
http://www.php.net
$link = mysql_connect('HOST', 'UID', 'PASS') or die (Could not connect);
mysql_select_db('DATABASE_NAME') or die (Could not select database);
$query = SELECT * FROM table;
$result = mysql_query($query) or
thanks for the reply!
i have installed [EMAIL PROTECTED] i need install more soft? odbc etc...?, i think if [EMAIL PROTECTED]
have inside the phpmyadmin i dont need more installed, this is true?
thanks
Vladimir
- Original Message -
From: Mark Ackroyd [EMAIL PROTECTED]
To: 'Asterisk
Doesn't heartbeat take care of this? It's been awhile since I've
configured it. If two servers join back together as master, one of
them shuts down its services. Maybe I'm just wishfully thinking..
There's also a directive to determine if a secondary should fail back
over to the master if it
Yes, I know, but I have I was think that heartbeat use status function in
init asterisk script to check if asterisk is alive, but status function is
for redhat.Are there any similar function in Debian?. And in respect of
slave, when slave get all resources and master wakeup, maste request for
Could you possibly use the redhat init scripts instead? Or at least
duplicate the functionality under Debian.
(I'm not too familiar with Debian, so I don't know how it does such things).
On 1/19/06, Tron [EMAIL PROTECTED] wrote:
Yes, I know, but I have I was think that heartbeat use status
Heartbeat monitors only 'life' of cluster members.
Service should be monitored by a custom script.
Mimmus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gary Richardson
Sent: Thursday, January 19, 2006 4:28 PM
To: Asterisk Users Mailing List -
Then, say you that I must to do a script that check if asterisk is alive?
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Mimmus
Enviado el: jueves, 19 de enero de 2006 16:51
Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Asunto: RE:
Thanks for all the posts everyone
So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I
would rather not have to apply patches just to get the two PCI cards to work
in the same box
The price difference between the cards you guys mentioned is interesting
I have also heard
On 1/19/06, Tron [EMAIL PROTECTED] wrote:
Yes, I know, but I have I was think that heartbeat use status function in
init asterisk script to check if asterisk is alive, but status function is
for redhat.Are there any similar function in Debian?. And in respect of
slave, when slave get all
Yes, heartbeat is good at monitoring system and network availability, but to
monitor applications as well, you need to jump through hoops and do some custom
development. A shame really because without that it's useless.
Also, heartbeat only works in a primary/secondary fashion. Ie you can't
Hello,
I'm trying to receive a fax to my inbound number from voipbuster.
Asterisk receives the call and starts the rxfax application successful,
but then nothing happens. The calling party is still hearing a ringing
tone, or sometimes nothing. Voicecalls are working correct and without
Can the # be used as a valid key press for a user in a dial plan?
if so how can the asterisk recognize it as a valid key press?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options
hello wath is wrong with this code???
$dbconn = mysql_connect(127.0.0.1,vladimir,vladimir);
mysql_select_db(cuentas);
$inDB = false;
//$link = mysql_connect(127.0.0.1, vladimir, vladimir) or die(Could
not connect);
//$db = mysql_select_db(cuentas, $link) or die(Could not select
database);
hi, im complete new with asterisk, so..
i want to be able using dtmf during a call, for execute a application.
Now i'm still making phonecalls trough a sip-adapter (Linksys pap2) with
my soundcart from server, and i receive in the asterisk-console putting
some digits from a analog-telephon:
Okay, sorry to hash out this discussion again, but it's starting to drive me
crazy
Successfully got the adapters to allow the BT phones to ring on lines coming
out of a TDM.. but now my latest problem is echo.
I have done tweaking of the gains in North and South America, and after a
bit
i have installed [EMAIL PROTECTED] i need install more soft? odbc etc...?, i
think if
[EMAIL PROTECTED]
have inside the phpmyadmin i dont need more installed, this is true?
I don't use [EMAIL PROTECTED] , but if phpmyadmin is installed you be pretty
sure that you
have all you need to
I don't know, I only tested it with a single TDM400.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tron
Sent: Thursday, January 19, 2006 3:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
On Thursday 19 January 2006 13:48, Joseph Rothstein wrote:
I am now getting these messages on a second box running a different version
of Asterisk. If anyone has any idea what is causing these, or how to avoid
them I would be very grateful.
157 Jan 19 11:59:50 NOTICE[6070]: Dropping
not sure if i get your point, but may be something like this:
exten = s,1,Answer()
exten = s,2,Background(dial_some_digits);
exten = _X.,MyApplication(${EXTEN})
where MyApplication can be a custom application, and ${EXTEN} is a
magic variable that will hold the dialed digits
regards
On
Chris:
I had the same problem and gave up. (Gloucestershire)
If I have the gains right down low (just enough so that the DTMF tones
are recognised), the echo is acceptable, but audio at the far end is
very low.
I think that forwarding my POTS line to the VOIP line is the only
sensible
Chris Earle (CBL) wrote:
So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I
would rather not have to apply patches just to get the two PCI cards to work
in the same box
Than don't use two Fritz! cards, but two hfc-s cards, or the 4-port
cards mentioned below.
The
$dbconn = mysql_connect(127.0.0.1,vladimir,vladimir);
snip
kinda looks ok, but I would seriously consider putting as much code as you
can into the dialplan rather than forking a PHP script for all the user
interaction.
For example, I use asterisk to handle loads of fax 2 email accounts. on 4
http://www.voip-info.org/wiki-Asterisk+config+features.conf2006/1/19, Moises Silva
[EMAIL PROTECTED]:not sure if i get your point, but may be something like this:
exten = s,1,Answer()exten = s,2,Background(dial_some_digits);exten = _X.,MyApplication(${EXTEN})where MyApplication can be a custom
Chris Earle (CBL) wrote:
I have also heard about BERONET isdn cards? a single Beronet 4-channel card
would suffice I think?
Yes. Beronet and Junghanns both have the same cards. (they just 'work'
different, junghanns uses zap interfaces, beronet mISDN)
So, as already mentioned, you have 2
On Wed, 18 Jan 2006, Kevin wrote:
Thanks for the reply Steve,
I was able to try what you suggested, but no, that did not solve the issue.
Now, I didn't think about this before (and this might sound dumb, but I
am new to asterisk) but the phones I am using right now are all on SIP,
so
On Thu, 19 Jan 2006, Andreas Sikkema wrote:
The same thing can happen when a SIP ATA is configured to
use rfc2833 but is also a little to lote with the filtering
out of the DTMF. So sometimes it's not Asterisks fault at
all ;-)
And then there's some IVR's that don't notice it at all,
It's mapped to blind transfer in features.conf -- If you want to use the
blind transfer feature, which I find easier than my phones' transfer
features, remap it to ## in features.conf. That way if you hit # it
dtmfs through to the target IVR, but you can hit ## real quick to get
the transfer
check irq supply on the * server -- When you run zttest do you maintain
over 98%? or like Steve suggested, the network may have congestion or
other errors ethereal may help you figure out.
I had a polycom 500 that was doing this to my user, 301s and 501s
wouldn't do it. Not sure if that was
Hi,
I'm having problems with the rxFax app. One of the messages that appear in
my console is:
Executing Set(SIP/something,
FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif) in new stack
-- Executing RxFAX(SIP/something,
/var/spool/asterisk-fax/1137692307.5.tif) in new stack
Jan 19 12:38:30
Hello,
some problems with transfer and zap...
one hfc-card in NT mode and one fritz isdn-card in server.
there is one gigaset SX353 isdn phone on the hfc-card.
anybody calls from external via capi and the call is bridged
to the zap-device. if you want to transfer the call via R-button on
the
Hey Steve and everyone,
I looked at the configuration, and unless I am missing something I don't
think they are configured
# ztcfg -vv
Zaptel Configuration
==
Channel map:
0 channels configured.
In the zapata.conf file, it is the sample version, but I didn't notice
Just look through the devices settings for suppress silence or transmit silence and don't supress or prevent transmission... this is a common problem inX-LiteRobOn 1/19/06,
Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
check irq supply on the * server -- When you run zttest do you
In article [EMAIL PROTECTED],
Dan Austin [EMAIL PROTECTED] wrote:
Tony wrote:
I should tidy it up and submit it, but haven't got round to it :-(
Let us know if you can. I'm already maintaining a grocery list
of patches to make MeetMe viable in my orginization, so one more
won't kill me.
Adam Robins wrote:
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
I designed a virtualized san and have been running it in production for
the last
Anybody knows if it's possible to disable zap echo cancellor from
dialplan only for certain outbound calls ??
I share the same phone lines for voice calls and faxes. Iaxmodem works
fine for me only turning off the echo cancellor, but I need it for
voice calls.
Any ideas ?
maxx
On Thu, 19 Jan 2006, [iso-8859-1] Michaël Gaudette wrote:
Executing Set(SIP/something,
FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif) in new stack
-- Executing RxFAX(SIP/something,
/var/spool/asterisk-fax/1137692307.5.tif) in new stack
Jan 19 12:38:30 NOTICE[12008]: channel.c:1906
On Thu, Jan 19, 2006 at 04:41:29PM +0100, Tron wrote:
Yes, I know, but I have I was think that heartbeat use status function in
init asterisk script to check if asterisk is alive, but status function is
for redhat.Are there any similar function in Debian?. And in respect of
slave, when
Massimo De Nadal wrote:
Anybody knows if it's possible to disable zap echo cancellor from
dialplan only for certain outbound calls ??
I share the same phone lines for voice calls and faxes. Iaxmodem works
fine for me only turning off the echo cancellor, but I need it for
voice calls.
On Thursday 19 January 2006 12:52, Massimo De Nadal wrote:
I share the same phone lines for voice calls and faxes. Iaxmodem works
fine for me only turning off the echo cancellor, but I need it for
voice calls.
Any ideas ?
IAXModem (and the device you're connecting to) should be sending out
/pipermail/asterisk-users/attachments/20060119/22
939450/attachment.html
--
Message: 7
Date: Thu, 19 Jan 2006 17:47:29 + (UTC)
From: [EMAIL PROTECTED] (Tony Mountifield)
Subject: [Asterisk-Users] Re: MeetMe Listen Only flag (|m)
To: asterisk-users@lists.digium.com
Message
I know this is a somewhat odd application, but we have a very good
reason for needing it.
Basically, I want asterisk to automatically redial a caller unless they
exit the system properly.
Here are some pertinate sections of the dialplan.
[AUTOBCSTART]
EXTEN=001,1,Meetme9${ENC}|pq)
I asked this a few days ago, and haven't gotten an answer (or seen my
message in the archive, yet). Since there were some email problems the
other day, I will just pose the question again.
I would like to know if there is a way to have a table, like zapata_conf
in a DB, and have asterisk
Adam Robins wrote:
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
http://www.coraid.com/ for a slightly different approach to large
storage capacity.
My php script called from my extensions.conf is working fine, however
each time I run it with agi debug mode on I see the message
AGI Rx Done
AGI Tx 510 Invalid or unknown command
which does not come from my script, apart from the Rx Done where the
'Done' is the message from my exit()
Hi,
I was only able to ChanSpy Agent
channels.
How do I monitor outgoing calls?
Thank youDov
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hello,
On Di, 2006-01-03 at 16:31 +0100, Giovanni Miano wrote:
Use meetme app
Unfortunately meetme is no solution for me. If nobody can help me, is
there at least anybody who has the same problem?
As far as I can see there are lots of people using the HFC PCI card, is
nobody using
Hi everyone and Steve,
Well the problem I wrote about is fixed. Here is what I did to resolve
the issue.
I was running kernel 2.6.11-1.1369_FC4smp before. I went and upgraded to
2.6.14-1.1656_FC4smp along with the development files (which I finally
found were not installed). After I
You could try removing the asterisk config, and zaptel config from boot.
and just placing it in the /etc/rc.local file
i did this for a while
modprobe zaptel
modprobe wcfxo. etc etc...
sleep 2
/usr/sbin/asterisk
Yes,
I did exactly that, but when I boot zaptel doesn't load wct1xxp.
Hi List,
I am working on least cost routing code on the moment, and I am
stumbling on a problem.
Say you have provider A having:
Prefix XXX0.10
Prefix XXXYYY 0.20
And provider B having
Prefix XXX0.15
You're stuck, because you cannot decide if provider B's
Red
alarm on PRI is a physical layer problem, as in your telco had an outage or
soemone unplugged the cable.
-Original Message-From: Dov Bigio
[mailto:[EMAIL PROTECTED]Sent: Tuesday, January 17, 2006 1:02
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] red
Hi.
DTMF recognition could be device problem.
For example I have a setup which confirm that:
However I stuck and don't know in what direction to continue.
Have such schema:
Asterisk - sipura3000 - Land Line.
When call comes from cellphone spa3k captures DTMF and in debug from
spa3k I see
Anyone know how to obtain firmware and starter .cfg files for Polycom
phones? Despite registering at the Polycom web site, I can't locate
this stuff.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
On Thu, Jan 19, 2006 at 10:37:49AM -0800, Steven Ringwald wrote:
I asked this a few days ago, and haven't gotten an answer (or seen my
message in the archive, yet). Since there were some email problems the
other day, I will just pose the question again.
I would like to know if there is a
On 1/16/06, John Falk [EMAIL PROTECTED] wrote:
Can someone show me how to set up DUNDi, I will be using it to connect
14 asterisk servers internally. I don't want to use it on the external
world. If anyone has any examples of connecting 2 or 3 (if their is a
difference) machines in a DUNDi
Polycom are analy retentive when it comes to this. You should be able to get
the older versions on their web site though.
Doug.
-Original Message-
From: Bill Michaelson [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 19, 2006 2:13 PM
To: Asterisk Users Mailing List - Non-Commercial
The O'Reilly TFOT book is full of errors. Two that pop into my head instantly
are it's referring to regcontext being able to execute dialplan commands upon
SIP registration and it's use of auth= in sip.conf in the DUNDi section. I
wouldn't trust it.
-Original Message-
From: Leif Madsen
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.9
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality
Tzafir,
Any suggestion about the chan.zap issue that doesn't load wct1xxp as module?
I don't know if you send the answer to my last e-mail, just our e-mail
server was under attack, and we lost a lot of e-mails.
Thanks
Carlos Alperin
___
--Bandwidth
The short of it:
I am unable to compile chan_bluetooth on Asterisk 1.2.1 on CentOS 4.2.
I installed using the [EMAIL PROTECTED] 2.2 iso. Server is a plain
Celeron 2.93GHz box. Asterisk source is in /usr/src/asterisk, newest
chan_bluetooth source is in
I am having problem with T1 configuration.
Following r the config..
Zaptel.conf
# Autogenerated by /usr/local/sbin/genzaptelconf -- do not
hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# It must be in the module
Hi,
I have a TDM400P running with only one FXO port running on a VIA
EPIA MS1 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it
hang and when I interrupt it with Ctrl-C that is the result: ¿anyone
have some idea about why isn't working?
Some additional info:
#
1 - 100 of 114 matches
Mail list logo
