Geoff Nordli wrote:
Hi Everyone.
On the Linux 2.6 kernel do I need to recompile the kernel in order to
compile the zaptel modules?
Using Mandrake 10.1 which runs standard a 2.6 kernel I did not have to
recompile the kernel to get them working.
cheers,
David
Thanks,
Geoff
On Wednesday 16 March 2005 17:27, Harald Milz wrote:
Kevin P. Fleming [EMAIL PROTECTED] wrote:
promise of the Opteron's potential :-) There are very cheap systems out
there (designed for workstations using Athlon64) and there are very good
systems out there, and the latter can be had for
My apologies to the rest of the readers for the flame, but Mr
Salvatore, you are sadly misinformed.
I like MySQL as well as PostgreSQL and they both have their merits,
but it's annoying to see someone give a recommendation of one over the
other based on ignorance instead of relevant facts.
If
Hi there,
I have a TDM400P (1 x fxs, 1 x fxo) which I'm attempting to run on linux
2.6 (gentoo), without much success at the moment. I have previously had
it working on a 2.4 installation, but when I moved to a new box and
installed a 2.6-based system, it failed to work. In both cases I'm using
Check the WIKI there is an example of how to do this very thing.
I implemented it on a customer a few months back and it works great.
http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config
- Original Message -
From: Luki [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Good Evening,
It seems that ChanIsAvail stopped working with the latest CVS, at least for
IAX2 channels
My dial plan hasn't changed, but the ChanIsAvail always goes n+101, same
dialplan works just fine with 1.0.7
Could anyone confirm that?
Regards,
Wojtek
snip...
exten =
On Thu, 2005-03-17 at 09:40 +1300, David Zanetti wrote:
ERROR[2215]: Signalling requested is PRI Signalling but line is
in Unknown signalling 896 signalling
ERROR[2215]: Unable to register channel '1-30'
WARNING[2215]: chan_zap.so: load_module failed, returning -1
Gianluca Colucci wrote on Wednesday, 16 March 2005 9:54 AM:
So, it seems to work: when I call the 98 extension I can hear the
music.
The problem is when I try to make some test with three other
extensions: I call one of these from my SIP client, the called
answers me and I try
Bugger. I knew I'd screw up the patch instructions.
Try this-
# cd to /var/build_aah/asterisk_src/asterisk
# patch -p1 path-to/apps-meetme-cbmysql.txt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Wednesday,
Hi Scott -
You are correct - this is in the CDR:
2005-03-16 16:38:32 IAX2/[EMAIL PROTECTED]:45 7607534720 Toll-Free Call
7607534720 Dial SIP/1022|15|rtm 2060 ANSWERED 00:28
So it looks like I am getting the caller ID to the phone, but all I am
seeing on any of the phones I have tested is
try the local channel.
the local channel allows you to have:
[default]
exten = 123,1,Dial(${DEVICE1},30,tr)
exten = 124,1,Dial(${DEVICE2},45,tr)
exten = 125,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
that will go thru the dial plan of 123 and 124. However, when I tested
it for what I
i'm trying to set up a 12sp+ and here's what I'm getting in asterisk console:
-- Starting Skinny session from 192.168.0.181
Device SEP00308062B5CD is attempting to register
Mar 16 19:06:35 ERROR[26875]: chan_skinny.c:1856 handle_message:
Rejecting Device SEP00308062B5CD: Device not found
Mar
At 3:44 PM -0700 on 3/16/05, Max W Blackmer Jr wrote:
Hello Everyone,
I would like to create an Intercom extension that will dial a group of
extensions which are connected to SIP phones. The SIP phones are setup
to auto answer a particular extension assigned to one of the lines in
the phone. All
Do some reading about contexts in *. Basically, you
want all "public" sip requests to land in a dialplan context that has no access
to PSTN, and requests from your own SER box(es) to land in another context (that
DOES have access to PSTN).
You can achieve this by adding an entry to your
Hi,
I'm able to see the callerid number, but not the name. I've tried your
suggestion by removing the quotes, but it doesn't help. Any
suggestions?
Thanks,
-Ryan
On Fri, 11 Mar 2005 17:53:17 -0600, Eric Wieling [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
I upgraded a DVG-1120M to a
I replied with the right answer but for the wrong post, Sorry about that.
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 16, 2005 6:35 PM
Subject: Re:
Is there anyway that i can add in a Zap device to be monitored? So far what
i can see is, its only monitoring SIP extensions.
At 05:23 AM 3/17/2005, you wrote:
Very cool ...
Have you tried to compile it against Mono?
-JB Hawaii
Thorben Jensen wrote:
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
Hi,
Strange problem that I can't seem to see any past postings on. New installation on OS X server. Voicemail ok from the handset. When an email address is added for MWI notification, asterisk crashes after the email is sent. Sendmail/postfix logs show no errors and the email is sent ok
OK. maybe i needed to fully stop/start asterisk...now i get
registered, but the phone says Program Update on the screen?
-- Device 'SEP00308062B5CD' successfuly registered
Requesting capabilities
Version Request
Received CapabilitiesRes
Mar 16 19:39:21 WARNING[27156]: chan_skinny.c:2301
well, answered my own question again: changed the version string as
you see below, comes right up... I would like to get a newer firmware
though!
; Typical config for 12SP+
[SEP00308062B5CD]
device=SEP00308062B5CD
version=P002D204; Thanks critch
;version=P002F204 ; Thanks critch
I just got mine working by commenting out the version.
;version=P002G204; Thanks critch
-Tyler
- Original Message -
From: Justin Richards [EMAIL PROTECTED]
To: Ryan Laginski [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Hello,
I need to know the following: when a call is answered, this call es sending
a to agent, but I need that the agent when receive the call in the desktop
appear all information of this number.
You can do this with the Flash Operator Panel. http://www.asternic.org
Regards,
--
Nicolás
I had the same problem with 7.4. Looks like the new
firmware is picker about the time server. With the old
release if the phone couldn't get the time it would
just use its internal clock. Now it does not display
time when it does not get an update.
I did fix the problem. Go into SIPDefault.cnf
On Wed, 2 Mar 2005 00:51:15 -0800 (PST), Alex [EMAIL PROTECTED] wrote:
Hi guys still have the problem to setup the IVR correctly.
I am forwarding call from ser :
if (method == INVITE) {
if (uri =~ sip:[EMAIL PROTECTED]){
log(1, Forwarding to Asterisk\n);
Huh, that sounds interesting. I never knew what the local channels
were for. I will give it a try. At least I know where to start now...
thanks C F.
--Luki
On Wed, 16 Mar 2005 20:01:32 -0500, C F [EMAIL PROTECTED] wrote:
try the local channel.
the local channel allows you to have:
Hi,
Was running an early version packaged for OS X which still has Voicemail1 in the load, which it defaults to.
Changing the commands in the extension list from Voicemail(u200) to Voicemail2(u200) has the correct package used and stable.
Cheers
Rob
Rob Gillan
Director
DZhoN Pty. Ltd.
[EMAIL
**# is the sequence to get into programing mode where you can setup
the network info. when in this mode use * for . and use # to
finish that setting (enter key if you will)
goto voip-info.org and do a search on 12sp for more details. i got my
12sp running in a couple hours after doing some
If you want the CallerID name, you can hack together a script to query
Google's phonebook via AGI. It's damn fast and you can cache the
results in a database. You don't always get a match, but in that case
you can resort to the city/state database from www.nanpa.com/reports
and at least get the
Hello List.
I am setting up asterisk as a central dialplan, voicemail and conference
solution, connected to 12 Cisco 1760 Routers running Call Manager
Express IOS distributed around the world. This is all done over VPN.
These routers all have PSTN access in their respective country.
So far
OK, great... the local forking approach works great. Example:
[extensions]
exten = 10,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
[test]
exten = 11,1,SetVar(_SIP_CODEC=g726)
exten = 11,2,SetVar(_ALERT_INFO=Bellcore-r6)
exten = 11,3,Dial(SIP/11,10)
exten = 12,1,SetVar(_SIP_CODEC=ulaw)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of David Uzzell
Sent: Wednesday, March 16, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Do you need to recompile the Linux 2.6
Hi guys!
I bought an device that it is
used to connect two PBX, two modems or two phones. It has two RJ11
connections.If you plug two phones and pickup one the other one will
ring.
I am trying to use this device
as a FXS to FXO converter with a X100P Clone. I plugged one end at the modem
Dan Austin wrote:
What errors are you getting?
In file included from app_meetme2.c:13:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE'
undeclared (first use in this function)
/usr/include/asterisk/lock.h:302:
I have installed asterisk 1.0.6 i am using xlite for testing.when i
transfer a call i get the music on hold when i put a user on hold using
Xlite i get no sound at all. dead airwhy is that, in the asterisk
log it is not even tring to paly the music on holdI have me
extention like the
Hi there,
Anybody on this list knows where I can obtain Hong Kong DID's from ?
Cheers,
Sahil
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Hello,
We 'll setup asterisk servers on several remote locations atleast 6-7
different locations these are connected to each other through
DXX(Digital Cross Connect) ,on larger locations we use PRI/E1 and
small locations we use TDM400 may be one or two but lot of IP
phones(soft/hard
Answering myself here, just thought of that I didn't put my version info
in there.
I'm running asterisk 1.05 on 2.6.9-gentoo-r13
Regards,
Maron Kristofersson
Maron Kristófersson wrote:
Hello List.
I am setting up asterisk as a central dialplan, voicemail and conference
solution, connected to 12
With what you are talking about, I don't think I'd find $125.00 for a
TDM10B outrageous. You could also plug a phone into your server ;-)
Maron Kristófersson wrote:
Answering myself here, just thought of that I didn't put my version
info in there.
I'm running asterisk 1.05 on 2.6.9-gentoo-r13
I'm using a little agi to get the name for *some* callers -
http://muware.com/asterisk
-Original Message-
From: Richard J. Sears [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 16, 2005 5:32 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] NuFone and CallerID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mohammed
Firdosh Nasim
Sent: Tuesday, March 15, 2005 11:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g.
WindowsMessenger) from different subnet
I need to deploy some quasi-virtual-PBXes, and I'd like to avoid having to
be hands on for each new phone number deployed... so I would like to set
up some administrative extensions that can record greetings... lets say:
[admin]
exten = 8(NXXNXX),1,Record($1|-greeting.gsm)
[incoming]
exten
I want to use a cordless (radio) phone as an extension on Asterisk.
When I dial the extension, there is no indication of ringing.
The ringing goes out to the base (I plugged a phone into the phone jack on
the base), but the base does not register ringing.
When I connect the base to PSTN, it works
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af John Breeden
Sendt: 16. marts 2005 22:24
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: SV: [Asterisk-Users] IPSwitchBoard BETA
Very cool ...
Have you tried to compile it against Mono?
who have been fabricated their own cards from Tormenta 2 PCI Card?
In Tormenta 2 PCI Card,I found that U1 is XC2S50-PQ208 from gerber files.
From google,I found that XC2S50-PQ208 is Xilinx family FPGA,
Would you like to give me VHDL code about this FPGA.
thanks a lot
On Thu, 2005-03-17 at 07:58 +0100, Thorben Jensen wrote:
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] P vegne af John Breeden
Sendt: 16. marts 2005 22:24
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: SV: [Asterisk-Users] IPSwitchBoard BETA
Hi Kong,
No, I have no support for monitoring of Zap devices at the moment. If there
is great demand for it, I will make it.
Thank you
Thorben
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] På vegne af Kong
Sendt: 17. marts 2005 02:33
Til:
William,
Thanks friend for your reply we have following setup:
ipphones--routerasterisk server with channel bank loaded with
Quad T1 card, also on channel bank several analog phones connected
we deploy these setup at our headoffice and other branch offices
situated in different cities we use
I tried wiki, but I got too many pages (I think all of them), ...as answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
bye
Ronald
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Thorben
Is source code available?? what license is it released under?
--
--
The source code is not available at the moment and the software is provided
as is - no warranties given :-)
tgj
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On Thu, 17 Mar 2005 15:18:02 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote:
I tried wiki, but I got too many pages (I think all of them), ...as answer.
I want to write an agi.
I need a HOW-TO, is there anything available?
It depends. What language do you want to use?
Chris
Adnan Ahmed [EMAIL PROTECTED] writes:
ipphones--routerasterisk server with channel bank loaded with
Quad T1 card, also on channel bank several analog phones connected
we deploy these setup at our headoffice and other branch offices
situated in different cities we use TDM400 cards.
I do believe that I saw that one. If I recall it occurs
if you run make from the apps directory. Change directories
up one level and run make.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dmitry
Melekhov
Sent: Wednesday, March 16, 2005 8:36 PM
Hi All...
I'm trying to get the intercom feature working on some snom 190 phones
but having no luck...
As you can see from the SIP trace below (from the called phone),
intercom=true is being appended
to the To: header as per requirements. I've email'd snom a few days ago
but have yet to
get a
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