Hi Michael,
I have a TDM400P on an Asterisk box with:
1) a FXO connected to the old pbx and
2) a FXS connected to a normal analog phone
3) the analog phone is a Telecom Sirio, (the most common in Italy)
If I knew how to check asterisk send/receive this non-digits signals it
can be easier to
Currently we have (with our NEC phone system) the options in voicemail to
have a message say press 2 to go to my mobile phone
Can this be done in asterisk without setting up an IVR for each user ?
Has anyone got a voicemail dialplan that can do this ?
Thanks
--
Kevin Withnall
ILB Computing
Asterisk has an option to have an out (by pressing '0') and you could
use that to jump out of voicemail and off to someones mobile.
Maybe a dbget to grab the mobile phone for the user would be a neat way
to go.
--
Paul Hales
Technical Manager
AsteriskIT
www.asteriskit.com.au
ph: 03 8320
Hi,
We're running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server
connected to the PSTN through two E1 pipes to a TE405P. This has been running
just fine for several months...
But yesturday we connected a large number of softphone SIP clients (50) and 25
of these where running
Devraj Mukherjee wrote:
Hello world,
Any success stories of getting a Nokia E61 to work with Asterisk
server? Interested to hear before we buy them for work :)
I don't know about e61, but I connected an e60 up yesterday that wasn't
any hassle.
Even the stories about poor quality with WPA +
Thanks for that, it works like a charm :-)
--
Kevin Withnall
ILB Computing
PH: 02 4227 0001 Mobile: 0412 453 846
FAX: 02 4227 0081
http://kevin.withnall.com/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Paul Hales
Sent: Tuesday, 4 July 2006
Greetings,
I have installed a new FXO card but even though there's no incoming route, it answers
the line after 2 to 3 rings. If I do create an incoming route, the same
happens, but it never rings the ring group or extension I enter. It's
almost as if the card acts as a modem. The caller
I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls.
Mvh,
Jan
-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 4 juli 2006 09:41
Till: asterisk-users@lists.digium.com
Ämne: [Asterisk-Users] Running 40
Hi all,
I have a Qsig link over a TE210P card between my asterisk box and a
meridian 81c which worked very well.
My problem is that no name is transmitted in both directions.
I always get messages like.
!! Unknown IE 49 (cs5, Unknown Information Element)
!! Unknown IE 50 (cs5, Unknown
Hi,
configuration for E61 is the same as E60.
As for the codec, G729 works between E60/61 phones (G729 passthru).
At 03:44 PM 7/4/2006, you wrote:
Devraj Mukherjee wrote:
Hello world,
Any success stories of getting a Nokia E61 to work with Asterisk
server? Interested to hear before we
Check the default context defined in zapata.conf which is where
incoming calls will go to. It may be going to a context that you are
not aware of.
- Daniel
On Jul 4, 2006, at 3:46 AM, Pierre du Plessis wrote:
Greetings,
I have installed a new FXO card but even though there's no incoming
Asterisk has some issues with the Nokia SIP client. I've started to
add some small
changes to svn trunk to support call hold with the Nokia, as well as
behave a bit
better in regards to ilbc encoding, even though that should still be
avoided.
I've had a lot of issues with the Nokia loosing
i am trying to sort out an issue with my SIP provider (I can make
outgoing calls but am not recieving calls) and have been trying to use
sip debug from the CLI. I am after a way to get these debug messages
into a file (I find it easier to go over a file than having to deal
with
all the
Hi,
I was wondering, after recording a call (through either the
monitor()-application or automon), is there a way to put the recorded
file into a user's mailbox? So far, we just send out the file as an
email attachment, but having it in my mailbox would just be so much more
convenient... (^_^)
On Tue, 2006-07-04 at 10:17 +0200, Marc Rohlfing wrote:
Hi,
I was wondering, after recording a call (through either the
monitor()-application or automon), is there a way to put the recorded
file into a user's mailbox? So far, we just send out the file as an
email attachment, but having it
They may chose to access the file via their phone. By delivering it to
their mailbox they have the option of either phone access or email
access (assuming voicemail is delivered to their email server).
Just a guess.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I don't understand what you are asking: what's the difference
between sending out the email as an attachment so it ends up
in a user's mailbox versus having it in the user's mailbox.
Aren't they the same?
Oops, my bad: I'm talking about the user's *voicemail* box here - should
have
I have trixbox 1.0 how I can update it to 1.1 or from where I
can download trixbox 1.1
Regards
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without
On 8/3/06, Khaled Chehab [EMAIL PROTECTED] wrote:
I have trixbox 1.0 how I can update it to 1.1 or from where I can download
trixbox 1.1
Have you tried the trixbox-update.sh script?
Mike
Regards
*
No
This is a good idea, good use of technology. You should be able to do
this by looking at the way voicemails are already being stored, just
add the file in and make the txt file with the relevant information.
Look in for hints:
/var/spool/asterisk/voicemail/context/mailbox/
On 7/4/06, Marc
On Tue, Jul 04, 2006 at 10:00:30AM +0200, Olle E Johansson wrote:
i am trying to sort out an issue with my SIP provider (I can make
outgoing calls but am not recieving calls) and have been trying to use
sip debug from the CLI. I am after a way to get these debug messages
into a file (I find
Khaled Chehab wrote:
//I have trixbox 1.0 how I can update it to 1.1 or from where I can
download trixbox 1.1 //
Do you want to get an answer?
Can you read at all?
Read the replies to your previous questions.
1. It is July not August. So fix your date.
2. Remove your disclaimer. It
Hello again,
I read this interesting article about the TE405P card. How do I check what
firmware version my card has?
http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html ... And how
do I update it if it's an old one?
Regards,
Jan
-Ursprungligt meddelande-
Från:
Mark, While this is a possibility, what I'm really looking for is some help in where to start debugging this problem.CheersOn 7/3/06, Mark Phillips
[EMAIL PROTECTED] wrote:Perhaps the BT crew are all on a drunken rampage along Sochiehall
Street?On Mon, 2006-07-03 at 15:14 +0100, Colin MacMillan
I think you'll need to set the jumpers on the card in order to specify
the NT ports.
Jean-Louis curty wrote:
Hi everybody
I hope that somebody can help me with the following
I have
2 quadbri cards
2 - 1t0 cards
1 pabx alcatel 4200
I would like to connect my asterisk to the alcatel
On 4 Jul 2006, at 09:58, Olle E Johansson wrote:
I've had a lot of issues with the Nokia loosing the registration
and WLAN access while
I'm still in the office. Anyone that have any remedies for that?
Yep, that's my main issue as well. I doubts it's a configuration
issue since there isn't
Hi,
You can call me by my first name (Silviu)
:))
I have made the changes to the settings file, I have
removed the LDAP-related settings - nothing changes... The file is still taken
into account, as other changes affect the phone, but the SIP fields stay
desperately blank...
I don't
Do you
know for sure that it's BT's announcement, and not one from your dial
plan?
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Colin
MacMillanSent: 04 July 2006 10:21To: [EMAIL PROTECTED];
Asterisk Users Mailing List - Non-Commercial
On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote:
I does nothing special,
no output, nor error,
same.. .:-(
you should at least get any output from ztcfg, but aside from that,
like Tommaso said, you must also set the correct jumpers.
cheers
___
Vidura,you would want to use some kind of IVR + php-agi to do the database operations (of course there are 10 other combinations - like Ruby - on -rails and RAGI). Quick suggestion: if you've played with asterisk before, I recommend that you look at
voip-info.org for php-agi links and
thanks to all of you, I fixed my problem by changing the cable !jl
2006/7/4, stoffell [EMAIL PROTECTED]:
On 5/31/06, Jean-Louis curty [EMAIL PROTECTED] wrote: I does nothing special,
no output, nor error, same.. .:-(you should at least get any output from ztcfg, but aside from that,like Tommaso
Hi Ken,I did a clean install of FreePBX n Asterisk with asterisk-addons, sounds. Have you downloaded perl and perl-CPAN? Check this link
http://aussievoip.com.au/wiki/index.php?page=freePBX-2.1beta1Install and scroll down to the section about mysql_addon-KimOn 6/28/06, Ken Chan
[EMAIL PROTECTED]
Hi C F,
I read the comments but the problem remains...after some tests, I
changed some parameters inside zapata.h and recompiled to make flash
button work so now my asterisk knows when the user presses the flash
button /during a call./
My problem now is how to transfer the flash signal to the
On Wed, Jun 28, 2006 at 03:42:45PM -0500, Ken Chan wrote:
Hello,
I am trying to install Asterisk-Addon and got the following problem:
-
cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o
cc -fPIC -I../asterisk
Thanks guys.
How about the quality of the call etc? Are you happy with the phone,
do you recommend them?
On 7/4/06, Antonio Rabena [EMAIL PROTECTED] wrote:
Hi,
configuration for E61 is the same as E60.
As for the codec, G729 works between E60/61 phones (G729 passthru).
At 03:44 PM
Hi,I just installed freePBX n Asterisk (Fedora 5, ast*1.2.9.1)and they are working well except when i created 2 extensions i.e n 1235, when i try to call either from my SIP Phones, when i pick the call from one of the extension, the call fails and i hear a ¨busy tone¨. Another problem arrises
On Tue, Jul 04, 2006 at 01:49:31PM +0300, Levis Kimotho wrote:
Hi,
I just installed freePBX n Asterisk (Fedora 5, ast*1.2.9.1)and they are
working well except when i created 2 extensions i.e n 1235, when i try
to call either from my SIP Phones, when i pick the call from one of the
Hi all,
I am running asterisk 1.2.9 + digium te110p
Does my setup about support outbound fax?
Regards,
rootlinux
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
On 7/3/06, Tristan [EMAIL PROTECTED] wrote:
Hi everybody !
I need to play a file to customers when an agent answered the line to
tell them it's their turn but I don't want to do periodic annoucements,
Is there a way or something I misunderstood in the voip.org docs because
I can't do this for
Good news !!!
Do you think I'll be able to use /trunk version of app_queue against
1.2.9.1 ? Or what (stable) version should I'll be looking to use this ?
Thanks,
Tristan
BJ Weschke a écrit :
On 7/3/06, Tristan [EMAIL PROTECTED] wrote:
Hi everybody !
I need to play a file to customers
So ...where can I get some help on my problem?thxchristian
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Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this configuration, and call
somebody
Looking (not so deep) at Your logs... mostly cdr INSERT INTO...
for me, it's looks that both calls was handled in [iax] context
so...
You specified [ivr-menu-number-context] - how do You make a jump to such
context ?
You have 3 dids 10,20,0 and 3 context,don't you ?
where dids patterns are
Hi,
I have a little problem related to quintum a400 gateway.
I have installed asterisk-1.2.8. Have configured it with SIP and H323
channels to recieve and make calls over lan using softphone (shphone
for both SIP and H323). H323 driver version is openh323-v1.17.1 and
pwlib-v1.9.0 . pc to pc calls
Hi Thomas,
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using
Your config files are not properly attached. i cant open them. send them again.On 7/4/06, Thomas Jacobsen
[EMAIL PROTECTED] wrote:Hello list,I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,and I hope some of you
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or point me in the
right direction. At the moment by using this
I can read them, so they are properly attached. Check You mail software.
Your config files are not
properly attached. i cant open them. send them again.
___
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To
Sorry I changed my text and deleted too much :-S
Kai Fürstenberg wrote:
Hi Thomas,
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me
Try Termilink. www.termilink.net
-- Original message -- From: "Carlos Chavez" [EMAIL PROTECTED] Now that Nufone is dead, what are other providers of 800 numbers that work with Asterisk? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México
Termilink, at www.termilink.net
-- Original message -- From: "C F" [EMAIL PROTECTED] Define best. On 5/23/06, Crazy Boy <[EMAIL PROTECTED]>wrote: Hi Friends, Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten = _9011.,1,SetCIDNum(XXX-XXX-|a)
exten =
Does phones are registered in Asterisk ? (CLIsip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this
I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make
install), there was no errors.
I used svn to get the polycom_acd_functions asterisk branch release 30432, I
have to run make 3 times as it as it comes up with making opts re-run make.
It then completes and I run make
Filip Drągowski wrote:
I can read them, so they are properly attached. Check You mail software.
Your config files are not properly attached. i cant open them. send
them again.
They are attached as:
Content-Type: application/octet-stream; name=extensions.conf
Content-Transfer-Encoding: 8bit
Hello,
It was a mistake(because i edited them before i send them to the list,
they are not original like that).
Best Regards,
Thomas
On Tue, 2006-07-04 at 15:19 +0200, Kai Fürstenberg wrote:
Sorry I changed my text and deleted too much :-S
Kai Fürstenberg wrote:
Hi Thomas,
Thomas
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or
Hi,Below is part of the log file
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Caller ID name is 'LAN201' number is '1235'
Jul 4 16:38:06 VERBOSE[5871] logger.c: dialparties.agi: Methodology of ring is 'none'
Jul 4 16:38:06 VERBOSE[5871] logger.c: -- dialparties.agi: Added
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me some advice or
Hi,
can anybody tell me where can i find help for configuring quintum gateway with asterisk?
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Hello,
It was a mistake(because i edited them before i send them to the list,
they are not original like that).
Best Regards,
Thomas
On Tue, 2006-07-04 at 15:19 +0200, Kai Fürstenberg wrote:
Sorry I changed my text and deleted too much :-S
Kai Fürstenberg wrote:
Hi Thomas,
Thomas
Are the phones behind a NAT? What is the processory memory size? Are the E1 channelized?
-- Original message -- From: [EMAIL PROTECTED] I should add that thease 25 calls where SIP (internal) to Zap (PSTN) calls. Mvh, Jan -Ursprungligt meddelande- Från:
Sorry I didn't realize this is how you wanted it to work - that the
user is on a FXS and you want when the user flashes that it flashes
the host pbx.
I disagree with you on this setup the user should be requried to press
some DTMF and not just flash the phone. The main reason being that
otherwise
Hello,
Yes all phones and trunks are registered.
/Thomas
On Tue, 2006-07-04 at 15:30 +0200, Filip Drągowski wrote:
Does phones are registered in Asterisk ? (CLIsip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a
Hi,
Von L. wrote:
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten = _9011.,1,SetCIDNum(XXX-XXX-|a)
i didn't thought of that, and i tried it - it works when i use the Goto commandbefore i had one incoming context like [iax] which includes the different sub-contexts ofthe three ivr-menus - and the menupoints of the first listet included context were played.
Interesting.thanks to you Filip !On
I've
encountered a strange problem in what I thought would be a
straightforward upgrade to Asterisk 1.2 and was hoping someone out here
may have run into something similar.
The system is Linux FC3 with a 2.6.9 kernel. The problem is that the
new wctdm module will not load during modprobe.
On 7/4/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote:
I have installed libpri 1.2.3 and zaptel 1.2.6 (with make clean, make, make
install), there was no errors.
I used svn to get the polycom_acd_functions asterisk branch release 30432, I
have to run make 3 times as it as it comes up with making
Filip Drągowski wrote:
Does phones are registered in Asterisk ? (CLIsip show peers)
CLI log showing such connections will be usefull (no debug for now).
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf).
I recently have been required to terminate traffic via H323. We have beensuccessfully handling this traffic as SIP. We often have 30 + concurrent calls on this server and I am not quite sure the best way to handle this
new H322 traffic. Which of the h323 channels for * can handle
I am having trouble setting up international dialing. I have an asterisk
server connected to a PRI at our collocation. I have this setup in my
extensions.conf file, yet I still cannot get connected to international
calls.
[OUTBOUND]
exten = _9011.,1,SetCIDNum(XXX-XXX-|a)
exten =
Hi Everyone,
I am new to Asterisk but I have found that quite a few people have implemented
it with the Mediatrix 1204. Does anyone know of a wiki or place where there is
good documentation regarding this configuration?
Thanks
Julian___
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I'm sorry for the double posts.
On Tue, 2006-07-04 at 15:52 +0200, Thomas Jacobsen wrote:
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf
Who says nufone is dead?
I use them, but my did is through sellvoip.net
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Hello,
I have been trying to get the SIP -- H323 working in the last few weeks. I
tried different H323 channel drivers. I need help badly.
I got SIP -- SIP (with canreinvite=yes) and it was working fine. So, I
believe the problem is not in SIP side.
Here are my problems:
a) I am currently
Hi,
I connected an Asterisk box to an old pbx using a TDM400P (one fxs and
one fxo). Then I connected an analog phone to Asterisk FXS port. Is it
possible to send a flash command to old pbx via asterisk box when
pressing flash button on the analog phone?
When I press the flash button the
Phones are not behind NAT.
Every client is on the sameinternal network as
the asterisk pbx (nothing is sent throughthe internet). It's not the
network since I tested this by calling asterisk from an outside phone (cell) and
let asterisk play a message for me. Same "cutting" and "chopping"
Thomas Jacobsen wrote:
On Tue, 2006-07-04 at 15:16 +0200, Kai Fürstenberg wrote:
Thomas Jacobsen wrote:
Hello list,
I'm a asterisk-beginner and could use some assistance with my
configfiles(sip.conf extensions.conf). I'll attach them to this mail,
and I hope some of you prof's can give me
On Thu, 2006-08-03 at 11:40 +0300, Khaled Chehab wrote:
I have trixbox 1.0 how I can update it to 1.1 or from where I can
download trixbox 1.1
Obviously the responses are falling on deaf ears so I'll just rinse and
repeat. Hopefully it will register this time:
1) trixbox questions should go
I just
resolved the problem.
The older zaptel kernel modules (IIRC) were installed into the /misc/
subdirectory and the newer modules are installed into /extra/. To
further complicate matters, I had entries in my /etc/modprobe.conf that
were still loading the previous kernel modules from the
Hello,
I decided to resend the files, because i made alot of typos in them. -
Please use these files instead.
Best Regards,
Thomas
extensions.conf
Description: Binary data
sip.conf
Description: Binary data
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Salve *!
I'm using asterisk for a while and now I want to have a colord CLI.
I have apt-get install asterisk/testing, that is asterisk 1.2.7.1
I use Debian stable/testing on a vserver with any /dev/tty*.
So, of course, I comment out #TTY=9 inside /usr/sbin/safe_asterisk.
/etc/init.d/asterisk
I want to get a variable, depending on the time.
I tried this one, but it does not work:
exten = 75,1,Set(guess=SYSTEM(echo $((1 + $(date +%S)*100 % 23)))
The idea is that the variable guess will change every 23 times per minute.
How would be the right syntax?
bye
Ronald Wiplinger
Have you tried without reinviting ?? (canreinvite=no)
Is your * box behind a nat ?
maxx
Scott Pinhorne ha scritto:
Hi All
I have setup my asteriks to use voipcheap.com for the outgoing trunk
on local calls (because they are free), my setup is below:
register = username:[EMAIL PROTECTED]
Joshua Laroff wrote:
I recently have been required to terminate traffic via H323. We have
beensuccessfully handling this traffic as SIP. We often have 30 +
concurrent calls on this server and I am not quite sure the best way to
handle this new H322 traffic. Which of the h323 channels for *
Hello List.I am looking to build an Asterisk Voicemail application to serve approx. 100 users.I will be building the Voicemail system using a standard Asterisk install on a stable Debian system.The system will house 100x20mb/each voicemail boxes.
On to my question:The Voicemail system will most
Phones are not behind
NAT.
Every client is on the sameinternal network as
the asterisk pbx (nothing is sent throughthe internet). It's not the
network since I tested this by calling asterisk from an outside phone (cell) and
let asterisk play a message for me. Same "cutting" and "chopping"
Hi C F,
ok, I also thought to make the user to press some keys for example * and
3 so I setup a little test made using an Asterisk box with a TDM400P (2
FXS + 2 FXO) connected to an analog phone (fxs port) and an analog line
(fxo port).
I searched on internet and found some interesting stuff
On Tue, Jul 04, 2006 at 10:06:27AM -0400, Jerry Brady wrote:
I've encountered a strange problem in what I thought would be a
straightforward upgrade to Asterisk 1.2 and was hoping someone out here
may have run into something similar.
The system is Linux FC3 with a 2.6.9 kernel. The
On Tue, Jul 04, 2006 at 10:53:46AM -0400, Jerry Brady wrote:
I just resolved the problem.
The older zaptel kernel modules (IIRC) were installed into the /misc/
subdirectory and the newer modules are installed into /extra/. To
further complicate matters, I had entries in my
On Tue, Jul 04, 2006 at 05:10:35PM +0200, Robert Michel wrote:
Salve *!
I'm using asterisk for a while and now I want to have a colord CLI.
I have apt-get install asterisk/testing, that is asterisk 1.2.7.1
I use Debian stable/testing on a vserver with any /dev/tty*.
So, of course, I
I am looking for a graphical statistic program.
What I want to see is:
a. my bandwidth (MRTG I use now from my upstream, but the time seems to
be 20 minutes wrong,...)
b. how many phone calls are at the same time (to get the feeling how
much bandwidth how many phone calls are using)
c. how
Hi JR,I also noticed this article and thought why not! let's try this. After I followed the document I still wasn't able to do dundi lookups. This is what I get when I try to do a lookup:Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: DPDISCOVER (Command)
Flags: 00 STrans: 31739 DTrans: 0
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Thanks for your email,
I am currently on annual leave and will return on the 19th July.
Many Thanks
Scott Pinhorne
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Thanks for your email,
I am currently on annual leave and will return on the 19th July.
Many Thanks
Scott Pinhorne
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Thanks for your email,
I am currently on annual leave and will return on the 19th July.
Many Thanks
Scott Pinhorne
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Thanks for your email,
I am currently on annual leave and will return on the 19th July.
Many Thanks
Scott Pinhorne
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Thanks for your email,
I am currently on annual leave and will return on the 19th July.
Many Thanks
Scott Pinhorne
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Asterisk-Users mailing list
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Thanks for your email,
I am currently on annual leave and will return on the 19th July.
Many Thanks
Scott Pinhorne
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