Hi,
whats is your problem ?
For me it works, but had problems too.
nicolas
Thorsten Gehrig wrote:
hi
anybody running with german SIPGATE?
my configuration don't works :-(
regards
[EMAIL PROTECTED]
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Title: Message
Dan,
This will
probably not work. Once Asterisk tells the Panasonic PBX to transfer the
call, the call will no longer go through the extension the Asterisk PBX is
attached to. It seems the only solution to this (doing it the way you are)
is to have two zaptel cards in the
Hi,
i want use both B channels on my isdn card (B1 ISA) but chan_capi open one
channel and asterisk say 2. channel is busy.
Must i use another isdn card ? I have a old B1 ISA card.
Can anyone help me with that ?
nicolas
SNIPS:
--
== DISCONNECT_IND PLCI=0x201 REASON=0x34a2
--
Hello i have the question:
is it possible to make a CFU like a isdn phone at the telekom it do ?
nicolas
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On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that you keep using, this is a
reminder to check into the
On Sun, 2004-05-09 at 14:33, Mark Elkins wrote:
On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that
Hello,
I manage a small office and we have a 4-year old legacy analog PBX
manufactured by Iwatsu. We have four incoming analog lines that terminate
to 7 different desktop phones.
The interface to Iwatsu requires Windows and the Iwatsu admin tools are
proprietary and not extensible.
What do I
Hi,
ztdummy says the following:
VoiceBOX:/usr/src/zaptel# modprobe ztdummy
/lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_unregister
/lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_transmit
/lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_receive
Ed,
Are you keeping the desktop phones, or upgrading to new phones?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ed Mansouri
Sent: 09 May 2004 14:00
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Where to start?
Hello,
I manage a small office and
Hello,
The goal would be to minimize the expenditure on new equipment, and
utilize as much of the existing equipment as possible, so I guess it just
depends on what the requirements of working with an open source PBX are.
The current desktop phones are pretty sophisticated and provide a lot of
http://bugs.digium.com/bug_view_page.php?bug_id=0001589
Has anyone else heard an audible blip, break or garble between answer and
the native bridge attempt using sip?
If I change the usleep(50); to usleep(5000); in rtp.c the proble totally
goes away... even the note above it says
Nope this was from my sipura ...
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Sunday, May 09, 2004 9:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 500ms usleep in rtp.c ?
Hi,
I'm trying to put together a simple gateway
configuration involving Asterisk. I have a machine
with 2 Digium X100P FXO cards installed and the
Asterisk Software, and I have 2 Sip Phones defined.
What I want to achieve is, any call arriving at FXO 1
is forwarded to Sip phone 1 only, and any
Hi experts (hope so),
I´ve behind a firewall (Linux FLI4L) - but i have configured all possible
Forwardings.
Two Problems at this time:
a) after many tries I have registered on siptel
*CLI sip show registry
Host Username Refresh State
217.10.79.9:5060 8003440
On Sat, Apr 03, 2004 at 02:36:59PM +1000, dkwok wrote:
Does any one regularly reboot GS101? It sometimes lost registration with
* and needs to be reboot.
What is the best way to do it by cron?
You might use curl for regular reboot as described in
Mark,
Would you please re-config or use a different mail client as to not send
your replies back as attachments??
It's VERY kludgy, and, I'm just going to stop reading them.. along with all
the other folks..
Thanks, Billy
- Original Message -
From: Mark Elkins [EMAIL PROTECTED]
To:
I´ve behind a firewall (Linux FLI4L) - but i have configured all possible
Forwardings.
Two Problems at this time:
a) after many tries I have registered on siptel
*CLI sip show registry
Host Username Refresh State
217.10.79.9:5060 8003440 120 Registered
Hi,
so i would do following:
1. you type your external ip into the sip.conf externip=x.x.x.x
2. nat=yes
Your forwarding are ok, think you need udp only : 5060 and 1-2 see
it in your rtp.conf.
And you need open your ports to.
should work.
nico
Thorsten Gehrig wrote:
Hi experts (hope
Bob,
What I am going to tell you may seem arrogant or what, but I think
you would do yourself a great favor if you figured this one out yourself
by studying the info that is available and ask questions if things don't
work. Your configuration is indeed very simple and with 100% certainty
you will
Hello,
From: Hermann Wecke [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stripping numbers at the end of a dial pattern =
extensions.conf
Date: 8 May 2004 22:03:57 +
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the
Hello all,
the scenario:
Carrier S2M-- * -S2M--Siemens
|
|
SIP Clients
and many other features
With much help from the list, the PRI links are without alarms and inbound
calls are
Hi,
Replying to my own mail. There is a mistake, The syntax is incorrect:
From: Girish Gopinath [EMAIL PROTECTED]
exten = 12345, 1, SetVar,MYDIGITS=${EXTEN:2,3} ; MYDIGITS = 2345.
Correct: exten = 1234, 1, SetVar,MYDIGITS=${EXTEN:2:3} ; MYDIGITS = 234.
My apologies...
Girish
Billy,
Attachment seems to be due to a GNUPG sig file
-- William
On Sun, 2004-05-09 at 12:00, Billy Huddleston wrote:
Mark,
Would you please re-config or use a different mail client as to not send
your replies back as attachments??
It's VERY kludgy, and, I'm just going to stop reading
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 07 May 2004 09:44 am, Uriel Carrasquilla wrote:
I have noticed that when I switched to macros in my extensions.conf,
there is now a 5 second delay.
The macro starts with an announcement and then voicemail.
Has anybody noticed the same?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
ANOTHER REMINDER NOT TO USE REPLY FOR NEW MESSAGES
Please note that this mailing list uses threading which allows us to track
each issue per thread.
When you press reply to send a message to the list it gets inserted into
someone elses thread, like
Some CVS upgrade in the last day or two has broken the recognition of
DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting
the error...
*CLI -- Executing VoiceMailMain(SIP/phone1-e0dd, ) in new stack
-- Playing 'vm-login' (language 'en')
**Here I push a button**
May 9
What firmware you have on that BT101? And yes gnupg or what ever you use to
sign your message did produce the attachemnt on this last one too.
bkw
- Original Message -
From: Mark Elkins [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 12:23 PM
Subject:
Mark,
Could you please add a SIP debug message with the SIP INFO?
/O
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I have used the cron job before, and it worked fine, but didn't seem to be more than a hack to me. I found that if I turned off the "Subscribe for MWI" flag in the GS config page, and it stopped losing registration. The MWI still works, it just stops sending the subscribe packets to the * box.
All:
Why do some requests to the manager return [--END COMMAND--] and some don't?
(version 0.7.1)
In the following example show version has it and sip show peers doesn't.
Why?
Thanks for any suggestions!
[Action: Command, Command: sip show peers]
[Response: Follows]
[Name/usernameHost
Hello,
i guess the problem ist pridialplan from zapata.conf
with
pridialplan = local
it works :-). But I still get the error messages:
Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
Presentation: Unknown (67) '' ]
I am trying to figure this out... I'm sure it's simple, but I can't
think of it right now
In my AGI Script I am doing this... (This is done in Perl)
$AGI-exec('Record',
/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav);
And after this is done.. I want to get the name of the file it created
PGPexch.htm.pgp
Description: Binary data
On Sun, 2004-05-09 at 20:48, Olle E. Johansson wrote:
Mark,
Could you please add a SIP debug message with the SIP INFO?
I've done a debug with a working asterisk (V1.0) and the non-working
asterisk. The trace is attached. :-)(debug - ascii text)
When you say SIP INFO - what else are you
Declare the file path before you record it.
$path = /usr/local/apache/htdocs/demo/sound/myapp.$date.wav;
$AGI-exec('Record',$path:wav);
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of AstGrp
Sent: Sunday, May 09, 2004 3:47 PM
To: [EMAIL PROTECTED]
Subject:
On Sun, 2004-05-09 at 21:39, brian k. west wrote:
What firmware you have on that BT101? And yes gnupg or what ever you use to
sign your message did produce the attachemnt on this last one too.
OK the gnuPG is off.. :-(
Product Model:BT100
Software Version: Program--1.0.4.63
That is not working...
I tried like you mentioned it and even a few different ways and will not
create the file at all
Am I doing something wrong...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09,
Ed Mansouri wrote:
Hello,
The goal would be to minimize the expenditure on new equipment, and
utilize as much of the existing equipment as possible, so I guess it just
depends on what the requirements of working with an open source PBX are.
The current desktop phones are pretty sophisticated and
I have the uhci_usb modules etc. installed.
Make sure this is a module and *not* part of the Kernel.
I have usb-uhci and usbcore as modules. What about PPP support?
Is that a problem? Should I also install it as a module?
regards,
thomas
___
Hi there,
today I made the German language prompts available for download:
http://www.karl.aegee.org/asterisk.nsf/HT/sound-de
Be aware: Asterisk doesn't yet fully support languages other than
English, there are still (smaller) issues with voicemail and date/time
announcements that require a
Never Mind... Figured it out... Thanks...
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
Lieberman
Posted At: Sunday, May 09, 2004 4:00 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] AGI Assistance
Subject: RE:
Can you post your error to the list so we know what was wrong?
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Sent: Monday, 10 May 2004 7:34 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] AGI Assistance
Never Mind...
Hi,
I've have followed through the help docs in trying to get an initial setup
going with two phones and the asterisk server. Firstly, all I'm trying to
do is get the two phones actually talking to one another VIA asterisk..
I've added this to sip.conf:
[phone1]
type=friend
host=dynamic
Title: Nachricht
Hello,
i have found a
webmin module on the astersik ftp server!
but how can i
install it?!
Can anybody help
me!
thanks in advance
best
regards
markus
Dohnal
Phone two is registering ok, double check phone one against phone two's
settings.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 3:51 PM
Subject: [Asterisk-Users] Help with initial setup
Hi,
I've have followed through the help docs in
Hi Felix,
on some UK public switches I have seen similar bad call setup problems with
a release cause of 28 (Invalid number format) when using:
pridialplan=national
Have you tried:
pridialplan=unknown
in zapata.conf?
It seems as though the omission of the pridialplan= statement in
Thanks Steve, that was good reading.
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 08, 2004 9:21 PM
Subject: Re: [Asterisk-Users] Low Bit Rate Codecs
Craig wrote:
Greetings all,
I have searched all over and have found
Thanks for that, the patch got me a little further.
I am now getting this error during the asterisk-oh323 compile.
Any ideas.
mipt:/usr/src/asterisk-oh323-0.6.1 # make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make[1]: Entering directory
On Mon, 10 May 2004, Administrator wrote:
i have found a webmin module on the astersik ftp server!
It is broken. Forget it.
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nasty. The higher rate (6.something kbps) sounds more reasonable. Using
30ms blocks, it is not so compatible with *, which is geared to 20ms
block processing. A lost packet causes a 30ms hole, so it tends to be
less tolerant of packet loss than something working in smaller blocks.
It
What kernel version??
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Schroeter
Sent: Sunday, May 09, 2004 4:24 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ztdummy problem?!?
I have the uhci_usb modules etc. installed.
Make sure this
Hello Again Felix,
first a quick apology: sorry, I re-read your e-mail and found the trace
information (lower down) that you had already posted. (It's late here, etc.)
The error messages that you reported in your last e-mail are actually
outbound Q.931 call setup messages that are being sent to
I've been trying to play the default music on hold file, but no luck yet.
here is my configuration:
extensions.conf
[incoming]
exten = s,1,Dial,Zap/2|10
exten = s,2,Voicemail,u34
exten = s,102,Voicemail,b34
exten = 34,1,SetMusicOnHold,default
Musiconhold.conf
[classes]
default =
It's obvious you've at least tried to figure it out since you've used
the SetMusicOnHold app, so I'll be nice. Try MusicOnHold()
http://www.voip-info.org/wiki-Asterisk+cmd+MusicOnHold
-Original Message-
From: leonardo [mailto:[EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 9:03 AM
To:
well, you have to have webmin installed first. you can get webmin from
www.webmin.com. Once you have that done, you can install the module
into webmin by logging into webmin with a browser, going into the webmin
management section then to the modules section and it become pretty
obvious
The error was in the Quotes and Date Variable
The end result looks as follows...
$Date = time();
$Path = /usr/local/apache/htdocs/demo/sound/$EmpNum.$Date;
$ShortPath = sound/$EmpNum.$Date;
$AGI-exec('Record', $Path:wav);
I needed the $ShortPath variable for some web values... But
Where i can download the module ??
regards
Ivan
-- Mensaje original --
From: Bruce Ferrell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk webmin
Reply-To: [EMAIL PROTECTED]
Date: Sun, 09 May 2004 19:30:33 -0700
well, you have to have webmin installed
um its in ftp.asterisk.org but its BROKEN
bkw
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 8:53 PM
Subject: Asunto: Re: [Asterisk-Users] Asterisk webmin
Where i can download the module ??
regards
Ivan
-- Mensaje
Look at bugs.digium.com - Search for Voicemail... You will find it
there...
Geoff Clark
Network Engineer
The Network Essentials
[EMAIL PROTECTED]
704-568-0031 (W)
704-622-3905 (C)
www.tnessentials.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Just downloaded the file caleld webmin.tgz. It appears to be a snapshot
of someones experimental stuff. It's mis-named to be able to install
per my instructions (my apologies all) and as structured webmin's module
installer won't. I'll see if I can get it into a form where it will
install
I have just installed one of the new TDM400 cards with an FXS and an FXO
module into my * server.
I also checked out the latest cvs head.
I am using 7940 phones.
Now I have some strange problems:
1. When in the VM menus, key presses do not register.
2. When I press hold on the 7940, it hangs
http://www.bkw.org/~brian/callingcard.conf
Now this is far from a complete example but it
doesn't allow more than 1 person to use the card at the same time(thats right
only one person at a time can use the card). Its just an example and
shouldn't be used in production. This is only to show
Its in CVS HEAD ... check /usr/src/asterisk/configs/voicemail.conf.sample
bkw
- Original Message -
From: AstGrp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 9:08 PM
Subject: RE: [Asterisk-Users] Voicemail: upgraded?
Look at bugs.digium.com - Search for
Mathieu Nantel wrote:
Hey,
Can anyone comment on the difference between the 7905 and it's upgrade, the
7905-G ? Has anyone used these phones in a configuration? Is SIP well
implemented in the 7905 ?
Thanks in advance,
Mat
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http://www.voip-info.org/wiki-Asterisk+cmd+StripLSD
gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Posted At: Saturday, May 08, 2004 6:04 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] Stripping numbers at the end of
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that you keep using, this is a
reminder to check into the configs/ directory of the Asterisk source tree, regardless
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