if you're on windoze, msn 4.7 works fine. v5 and v6 doesn't, though
roy
On Mon, 2003-07-21 at 12:30, Stuart Hirst wrote:
Does anyone have any views on the best software base SIP client to use
that normal users could use with Asterisk without being too techie ?
I have tried the X-Lite
Hello,
Anybody experience this error:
ERROR[237594]: File chan_h323.c, Line 875 (create_connection): Unable to
allocate private structure, this is very bad.
the call still get through, but both party cannot hear each other
Pls Help.
Foong
- Original Message -
From: Steven Critchfield
Hi,
What the Call Format means in IAX?
For example:
-- Format for call is 2
It is codec related?
Thanks,
Dan
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Hi,
This page does not exist...
Thanks,
Dan
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 25, 2003 8:38 AM
Subject: RE: [Asterisk-Users] time and date stamp in voicemail
Try looking drunkencoder.com/asterisk
On Thu,
Granted it takes some pre-thought, but why
not just use the *72/*73 codes to forward
when you are out of the office?
This worked out pretty well, I got to meet lots of the
other officemembers' family and friends! Ha ha!
Seriously, I think Asterisk can do better, just need
to figure out
Granted it takes some pre-thought, but why
not just use the *72/*73 codes to forward
when you are out of the office?
This worked out pretty well, I got to meet lots of the
other officemembers' family and friends! Ha ha!
Seriously, I think Asterisk can do better, just need
to figure out
On Fri, 25 Jul 2003, Kelvin Chua wrote:
yes, i agree, we never really felt the need to use unity, *'s vm is
functionally ok with callmanager
(except for the message waiting indication, or is there?) can *'s vm send a
MWI to the callmanager?
Not yet.
However, CCM supports MWI notifications
On Thu, 24 Jul 2003, Daryl Jones wrote:
Brad's recent list of enhancements look good, but I haven't looked
at the code yet. If the code looks good, I hope it will be committed
to the project CVS.
Thanks for the vote of confidence but I fear it's premature. Hopefully
someday I can make it
Title: Message
Hi,
Do you
still have it for sale ?
Abdul
Hakeem
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kalin
DikovSent: 21 July 2003 20:14To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] URGENT!
brandly new Wildcard E400P
Hi.
Is it posible to put a Asterisk server behind a
NAT-firewall, and let it be reachable from
the Internet..
Like this:
asterisk - nat - inet -
SIP
or am I forced to have the Asterisk box
connected
"directly" on the internet?..
Than
Hi all !
What is the current status of the Dialogic channel driver ?
Is it available ? Is it commercial ?? Any info ?
Thanks
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I asked the same question a couple of weeks ago and was told by Digium that its not
commercially available yet but the source code is available under NDA with Digium.
I'll dig out my contact and send off-list ...
Adam
-Original Message-
From: Marcel Prisi [mailto:[EMAIL PROTECTED]
It probably can be done but you would have to forward a number of ports from the NAT
to the * box..
Also be carefull of double NAT situations..
ie. Asterisk--NAT--internet--NAT--UA
becasue this is a real headache to get working..if you ever do..
Later
Hi.
Is it posible to put a Asterisk
I am trying V2 now but having problems. I am having the same issues I
had with build 1016 in that the call is set-up OK but no voice path in
either direction. I guess this is one of the setting in V2 1016 but I
have not worked out which one yet.
Stuart
-Original Message-
From: [EMAIL
Hi all,
is it possible to go on in the context after the dest channel hung up?
For example:
exten = 111,1,Dial,Zap/4
If the originating channel is connected to Zap/4 and the destination channel
(Zap/4) hangs up, both channels will be destroyed.
Is there any option or whatever for
Hi,
we want to use asterisk as a replacement for our current pbx but before
changeing we want to make some tests.
Therefor I have a PC with an AVM-A1 ISDN4Linux hardware and inside the
network a cisco 7960 Ip Phone (SIP) and two TIPTEL innovaphone 200 (H.323)
and some softpgones like kphone,
Hello everybody,
I have installed Asterisk from CVS (18/07/2003) and although everything
works fine, SetLanguage application doesn't seem to work. As it used to
work with previous version I wonder if I am missing something here.
The relevant line in extensions.conf is:
exten =
Does anyone have X-Lite v2 Build 1047 (the new one) working ?
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stuart Hirst
Sent: 25 July 2003 10:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Best software SIP client
I am trying V2 now
On Thu, 24 Jul 2003, Jeremy McNamara wrote:
Siggi Langauf wrote:
Are you running CCM 3.3(2), too?
No idea, I avoid dealing with CCM at all I fought tooth and nail to
stop them from wasting money on it, but they wouldn't listen to me.
Same thing here: they're probably going to pour
I've previously been using G711alaw on both the AS5300 and the phones but feel the
need for a less bandwidth hungry codec for those users that are connected behind ADSL
and so was investigating G.729 but ..
Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940
hello,
sometimes my capi_channel stop works - e.g. when i try to call number
which does not exist ( typo error ) and i must restart asterisk.
following lines appears in the log files :
ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free
channel on controller 1! will continue
upgrade to 0.2.4 :)
On Friday 25 July 2003 14:09, Marian Danisek wrote:
hello,
sometimes my capi_channel stop works - e.g. when i try to call number
which does not exist ( typo error ) and i must restart asterisk.
following lines appears in the log files :
ERROR[393234]: File chan_capi.c,
Siggi Langauf wrote:
No idea, I avoid dealing with CCM at all I fought tooth and nail to
stop them from wasting money on it, but they wouldn't listen to me.
Same thing here: they're probably going to pour Millions of Euros into
the Cisco dump during the next year. Unless I manage
Holger,
Therefor I have a PC with an AVM-A1 ISDN4Linux hardware and inside the
network a cisco 7960 Ip Phone (SIP) and two TIPTEL innovaphone 200 (H.323)
and some softpgones like kphone, netmeeting, ...
[..]
Does anyone have such a configuration in use and can send me the
configs? This would be
Tools | Options | Accounts
Choose a communication service and type your username
click 'Advanced'
type the name of the server and choose 'udp'
press ok until you're out
open regedit
browse to HKEY_CURRENT_USER\Software\Microsoft\MessengerService
change CorpPC2Phone to 1
play
On Friday 25 July
I am having the same probs. I get local dialing tones but no audio after the call is
connected.. I got a private build from Xten and it was the same
Dave
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Steven Thomas wrote:
Michael,
my mistake - more testing confirmed that the wrapper did not update in the
correct location. Asterisk was still using 0.5.3. Replaced with 0.5.4 and
the call is no longer dropped.
Asterisk still reports H.323 CONTROL PROTOCOL ERROR (Roundtrip Delay) but
it
Fixed it I have audio now... uninstall everything xten makes and manually clear
out all the xten/xlite stuff from the registry.. search for XtenNetwork and kill the
keys. reinstall Xpro and it works... go figure
Dave
___
Asterisk-Users
Do 'iax2 debug' to see more.
Martin
On Fri, 25 Jul 2003, Richard Scobie wrote:
A call is placed via IAX2 from one asterisk to another, to a TDM400
channel whose extensions.conf entry is
exten = 502,1,Dial(${COLIN})
exten = 502,2,Congestion
If this channel is already busy when called,
Any one able to tell me where this can be downloaded form?
I only see the generic download link - http://www.xten.com/download.php but this seems
to be version 1
Thanks
Damian
From:Dave Packham [mailto:[EMAIL PROTECTED]
Sent:Fri 25/07/2003 15:38
What Steve says... Also, check your hosts file for strange entries. This
was the problem in our case. We had exactly the same symptoms with RedHat
8.
-wade
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven Critchfield
Sent:
Thanks Roy, I this worked! Only one thing I can't seem to do- If I have
a password set in my sip.conf as in the 'secret' key, I can't get the
msn client to authenticate properly. (And yes, I'm typing the exact
same word I have in secret =)
Neel
Tools | Options | Accounts
Choose a
Dave
I tried this and I still have the same problem. I am using X-Lite though
and not X-Pro.
The SIP registration is fine but still no audio.
If anyone has X-Lite either 1016 or 1047 (v2) working, please could you
let me know and maybe email your registry settings for the app.
Stuart
Try http://brands.xten.net/x-lite/download/X-Lite_Install.exe
-Original Message-
From: Asterisk Maillist [mailto:[EMAIL PROTECTED] On
Behalf Of Asterisk Maillist
Sent: 25 July 2003 16:10
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Best software SIP client
Any one able to tell me
I had this issue until I fixed the DNS resolver on the * box.
Asterisk was attempting to deliver the mail message and having to
timeout name servers, etc. Once dns was setup properly for the box, the
message was delivered instantly and there was no more delay.
Now a good fix would be the spawn a
Hi Holger,
the AVM-A1 (fritz card) has capi4linux drivers (ftp.avmd.de/cardware)
which may already be included in your distro.
get chan_capi at http://www.junghanns.net/asterisk/ and read the
README and INSTALL file.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite
Stuart,
X-Lite v2.0 build 1050 was just released. Try that.
http://brands.xten.net/x-lite/download/X-Lite_Install.exe
Cheers,
Erik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stuart Hirst
Sent: Friday, July 25, 2003 10:12 AM
To: [EMAIL PROTECTED]
Hey Neel,
On Fri, Jul 25, 2003 at 10:40:55 -0500, Neel Datta wrote:
Thanks Roy, I this worked! Only one thing I can't seem to do- If I have
a password set in my sip.conf as in the 'secret' key, I can't get the
msn client to authenticate properly. (And yes, I'm typing the exact
same word
Dan,
the page is actually http://asterisk.drunkcoder.com/patches/ . However, I
didn't see the patch there.
Sincerely,
Andy Hester
Consero
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Sent: Friday, July 25, 2003 2:52 AM
To: [EMAIL
All
I have given the * PHP web interface files to Mark to check out. hopefully he will
include them into the CVS tree soon.
Dave Packham
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If asterisk is running as a daemon (started with no
command line options), can I reconnect to it and get
the vvverbose info in the console? In other words,
will
asterisk -rvvvc
work? Sorry to ask I would just try it to see if
it works, but I don't have an installation handy and
have to do
Erik,
Thanks for the info.
I have tried build 1050 on two different PC's and still the same
symptoms.
One the main machine I have been using for testing I removed the old
version, cleaned the registry and installed build 1050 but still no joy.
This is a Dell notebook in a docking station.
The
Yes, you can start this way and get most of the call flow detail just like
when you connect on the main screen. However, if you are writing your own
AGI scripts, you wouldn't get any output directed to STDERR (like debugging
messages) - these go only to the initial console where * is started.
It is selected by jumpers on the card. You may override also by using
t1e1override=foo when you modprobe wct4xxp
Mark
On Thu, 24 Jul 2003, Alex Lopez wrote:
One a t400p I know that I have 24 channels per port for a total of 96. However the
T410 card allows for E1 as well as T1 lines. How
I am also having the same problem with x-ten. I have tried release 1050
and it still does not work (no audio stream).
Steve Besch
Erik Lagerway wrote:
Stuart,
X-Lite v2.0 build 1050 was just released. Try that.
http://brands.xten.net/x-lite/download/X-Lite_Install.exe
Cheers,
Erik
Martin Pycko wrote:
Do 'iax2 debug' to see more.
Martin
Thanks Martin,
As it seems as if it may be a bug, I'll get IAX2 debug output from both
*'s and put them in the bug tracker to save list clutter.
Richard
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I can't seem to get musiconhold to work. I'm running asterisk on a RH9
box, I have the mpg123 package installed. In my zapata.conf file I have
the line MusicOnHold=default . In my musiconhold.conf file, in the
classes section I uncommented default and loud. In my extensions.conf
file I
John
Thanks for the response. This seems to be what I am looking. However, I
have discovered a problem with a simple perl script triggered from the h
extension.
I am using perl-Asterisk and if I call the script from any extension in
works fine. However, if I call the same script from h the
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with get_var) is gone. pass the
channel variables you need to acces as an argument to the agi script,
e.g.: exten = h,1,AGI(myagi.agi,${EXTEN} ${CALLERIDNUM})
regards
kapejod
--
I just got build 1050 working I had the same problem until I set Send
Internal IP: on in the menu under sip proxy
Kyle
- Original Message -
From: Stuart Hirst [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 25, 2003 1:05 PM
Subject: RE: [Asterisk-Users] Best software SIP
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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On Fri, 25 Jul 2003, Dave Packham wrote:
Fixed it I have audio now... uninstall everything xten makes and
manually clear out all the xten/xlite stuff from the registry.. search
for XtenNetwork and kill the keys. reinstall Xpro and it works... go
figure
For what it's worth, I was having
Thanks for the response. In addition to what you stated, I think there is
another problem with Asterisk::AGI
This is the test script
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $num = $AGI-get_variable('FOO')
Did anybody figure out how to make dial detect a busy on a zaptel channel on a
pri interface when using overlap dialing? According to the documentation dial
should return to priority n+101, if the called party is found to be busy. I can
see a DISCONNECT message with user busy coming from the
On Wed, Jul 23, 2003 at 10:56:47PM -0400, Jeremy McNamara wrote:
Either don't use unity and point all the IP phones to * for VM or just
setup an H.323 trunk that dumps u into the appropriate mailbox, if u
must use it.
This is what I'd like to see happen. I'm slowly but surely getting
is there any way to keep those vars around until after h goes away?maybe move the
free routiene to after h is done?
Dave
[EMAIL PROTECTED] 7/25/2003 5:32:55 PM
Hi Dan,
no wonder. when the h extension is called the channel (including all
the channel variables you want to read with
On Friday 25 July 2003 14:12, Andy Hester wrote:
Dan,
the page is actually http://asterisk.drunkcoder.com/patches/ .
However, I didn't see the patch there.
I just added it. It's available there now. Note that there are three
files: a patch, sounds, and some instructions.
-Tilghman
Tilghman,
Thanks alot for posting that. I'll check it out
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman
Lesher
Sent: Friday, July 25, 2003 10:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] time and date stamp
Tilghman Lesher wrote:
On Friday 25 July 2003 14:12, Andy Hester wrote:
Dan,
the page is actually http://asterisk.drunkcoder.com/patches/ .
However, I didn't see the patch there.
I just added it. It's available there now. Note that there are three
files: a patch, sounds, and some
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