Can I use queuemetrics with asterisknow?
I mean, if I modify the dialplan to use queuemetrics (I still don't
know if it's possible), will I loose my changes when the time comes to
do a conary update of the asterisknow package?
thanks,
--
2007/5/21, 0xception [EMAIL PROTECTED]:
the software was unstable and crashed a lot (at least the Linux version
was) ...
Which version did you then use ?
Regards
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asterisk-users mailing
Hello,
I'm looking to do the following, and I wonder if Asterisk can be used for it,
and if yes, if anyone can point me to the relevant information (commands,
sample config...):
1. Caller dials 111, 222 or 333.
2. Based on the dialed number, Asterisk queries an external MySQL table and
David,
Have a look at:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
-- Alex
--
Alex Balashov [EMAIL PROTECTED]
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To UNSUBSCRIBE or update options
in 1.4, func_odbc is your friend.
Julian.
David wrote:
Hello,
I'm looking to do the following, and I wonder if Asterisk can be used for it,
and if yes, if anyone can point me to the relevant information (commands,
sample config...):
1. Caller dials 111, 222 or 333.
2. Based on the dialed
Estarei ausente do escritório a partir de 21/05/2007 e não retornarei até
11/06/2007.
Responderei à sua mensagem quando retornar.
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hello, asteriskers:
i compile asterisk 1.2.18 in arm-linux. i got this error :dlfcn.c:40:
mach-o/dyld.h: No such file or directory. i check the /usr/include dir, there
is no mach-o dir and dyld.h file in /usr/include. i think i am missing
somethings in the cross-compile tools. Does nayone know
I have figured out a way to include dialed number in recorded
voicefile in freepbx . You have to edit
/var/lib/asterisk/agi-bin/recordingcheck
add this lines after $agi=new AGI()
$temp= $agi-get_variable(DIAL_NUMBER);
$agi-verbose(Number to be dialled is -{$temp[data]});
After this you can use
I realized that queuemetrics uses Java.
Is java available as an rpath package or do I need to get it from sun?
Also, will it break asterisknow?
Thanks.
On 5/21/07, Erick Perez [EMAIL PROTECTED] wrote:
Can I use queuemetrics with asterisknow?
I mean, if I modify the dialplan to use queuemetrics
Func_odbc is actually also backported to 1.2, so its your friend there too.
Regards
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 21. maj 2007 08:59
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi I'm looking for some help with Vicidial, If you have experience with
it and could help with some consulting please contact me off list.
Cheers,
Joel Hill
Asterisk IT
[EMAIL PROTECTED]
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... but when I pick up the handset, I get no voice either way, even
when I set the Linksys gateway to use a static external IP address
(STUN doesn't seem to work).
asterisk doesn't do STUN AFAIK, but I've never needed it and I use
double NAT and have since 1.0.?.
What happens when you do the
Hi,
I am trying to configure asterisk to translate between rfc2833 and
inband DTMF.
I have a cisco gateway which is configured as a trunk, and a cisco IP
phone which is registered to asterisk. The gateway does not support
rfc2833 and the IP phone does.
I tried changing directrtpsetup to no, and
I believe it was a version or two ago... I just downloaded the openWango
software again (current build) and it hasn't crashed on me...
again I have not done any sort of extensive testing.
On 5/20/07, Olivier [EMAIL PROTECTED] wrote:
2007/5/21, 0xception [EMAIL PROTECTED]:
the software was
Friday May 25th 2007 12:30 PM EDT Asterisk Users Live Conference/Podcast
Here's a chance to ask Kerry questions about trixbox.
See http://x2z.eu for access information.
Listen: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
Friday June 8th 2007
Stefan Wintermeyer, author of a soon to be
Hi,
I need to share my PSTN line with my Digium card together with my FAX machine.
If fax coming in, will asterisk pick up the call or my fax machine pick up the
call.
How do I make asterisk not to answer the incoming fax and let my fax machine
receive
the fax. Similarly, how do I make my fax
Hi there,
Just to announce that I've improved upon a greasemonkey script which allows
users to dial any number (in the given regex format) by turning it into a
clickable hyperlink.
The script uses greasemonkey's ajax callback to a simple php controller
script, so that the click does not
Hi, folks:
Is there any reason why MusicOnHold() would die after 60 seconds? That
looks suspiciously like a default timeout. How can I make it indefinite?
-Stephen-
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asterisk-users mailing list
Stephen Bosch wrote:
Hi, folks:
Is there any reason why MusicOnHold() would die after 60 seconds? That
looks suspiciously like a default timeout. How can I make it indefinite?
Moral of the story -- don't work at 4 am.
The call terminates after 60 seconds because I never answered it.
Hi!
Either the fax machine or the asterisk box has to pick up the call to
know whether it is a fax or not.
My solution is that I let asterisk pick up every call, and if it is a
fax, then the call is forwarded to a fax-machine.
If its a voice call, the call is forwarded to the phones.
Hello,
Please post to the jobs/consulting forum on the VICIDIAL forums site:
http://www.eflo.net/VICIDIALforum/viewforum.php?f=6
Thanks,
MATT---
On 5/21/07, Joel Hill [EMAIL PROTECTED] wrote:
Hi I'm looking for some help with Vicidial, If you have experience with
it and could help
Hi,
Thank you very much. I will test your method
ASLAY
- Original Message -
From: Thomas Artner [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 21, 2007 6:49 PM
Subject: Re: [asterisk-users] asterisk
Hi,
I was looking for a way to pass alphanumeric variables to asterisk via
the keypad, found this application app_dtmftotext.c , And I already tried
with 'spandsp' application for this. But I am getting errors.[I followed the
instructions at http://www.soft-switch.org/installing-spandsp.html]
I was looking at the ILECs' web sites to determine how their users access
voicemail.
I looked at ATT, Verizon, Qwest, and Embarq.
They supported one or a combination of the following for calling from your
phone:
*98
#55
Toll free number
Your number
A varying phone number, based on
Is there any FireFox plugin that contains an entire (SIP or IAX)
softphone, that can also be scripted in the page's HTML/Javascript?
On Mon, 2007-05-21 at 06:20 -0700,
[EMAIL PROTECTED] wrote:
Date: Mon, 21 May 2007 10:51:09 +0100
From: Richard Hamnett [EMAIL PROTECTED]
Subject:
I really appreciate your help :-)
On Mon, 21 May 2007 10:15:40 +0200, randulo [EMAIL PROTECTED]
wrote:
What happens when you do the echo test, call it from each phone?
Cool, I didn't know about Echo() .
I added extension 111 from this example:
http://www.asteriskguru.com/tutorials/echo.html
Hello,
I want to delete the voicemail messages that are in the Old
voicemail directory, 7 days after the listening of the message by the user.
Is someone as an idea how to do that???
Thanks.
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Really Great!!! Works for me in France I have just change the pattern
and that's ok reallygood job!
Cheers,
Alex
Richard Hamnett a écrit :
Hi there,
Just to announce that I've improved upon a greasemonkey script which
allows users to dial any number (in the given regex format) by turning
Thanks to all that have helped me so far. I have made a lot of
progress. I am able to make prilib and zaptel. Now to Asterisk...
After installing the kernel source, I have:
# cd /usr/src/linux
# make cloneconfig
# make prepare-all
Then I have run ./configure in the asterisk-1.4.4
On May 19, 2007, at 5:17 PM, Deepak Naidu wrote:
I think the best way is to conact Digium Hardware support. it seems
there may be an IRQ problem.
No, that doesn't have anything to do with IRQ problems. It looks like
it's another problem.
Matthew Fredrickson
--
Deepak
Francois
Carlos Chavez wrote:
On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote:
Use the cdr's, who wont know who but at least which phone did it.
I tried following the CDR but if I dial extension 4000 and extension
4002 picks up the call using *8 the CDR says that extension
I purchased a Grandstream 4 FXS Gateway and my * extensions are
not working. I disable the special features and changed the DIAL to
{X*#+} but not luck.
I can dial any other number, receive calls and so on. This is the
only thing that seems to be an issue.
Has anybody found a way around
On Sat, 2007-05-19 at 12:12 -0400, Steve Totaro wrote:
I have no affiliation with them but if their quotes are accurate then
they provide quite a few options as far as TDM connectivity and realtime
pricing.
If you do not want a phone call from a sales person, give them a BTN
that goes to an
I had to re install the my Asterisk BE with the latest version, and when I try
to load my g.729 codec license I do not see the folders in the path that they
are described in the instructions given to us with the license or in your
online documentation. I installed the disk 1 immage (rPath),
Another solution:
http://www.vikingtelecomsolutions.com/catalog/model_FAXJ-300.htm?sid=046EBF6027C7A0D38E77EAF75B184540pid=1209
Jorge
aslay-pinwee wrote:
Hi,
Thank you very much. I will test your method
ASLAY
- Original Message -
From: Thomas Artner [EMAIL PROTECTED]
To:
On Mon, 2007-05-21 at 12:49 +0200, Thomas Artner wrote:
My solution is that I let asterisk pick up every call, and if it is a
fax, then the call is forwarded to a fax-machine.
If its a voice call, the call is forwarded to the phones.
That is what I do as well. Use the fax extension to forward
Matthew Rubenstein a écrit :
Is there any FireFox plugin that contains an entire (SIP or IAX)
softphone, that can also be scripted in the page's HTML/Javascript?
Have you looked at MozPhone (http://moziax.mozdev.org/) ? It's a Firefox
VoIP extension IAX softphone, and Asterisk manager
I was able to fid the modules directoty, but when I run
-r-x-- 1 root root 1288344 May 21 11:35 register
/root/register
I get the following error
-bash: /root/register: cannot execute binary file
I have changed the file attributes as you can see on the ls -l
Hi,
My asterisk server was working with a 4-FXO analog
card (TDM400P).
I recently added two digital cards: a TE120P (1 PRI)
and a B410P (4 BRI).
The B410P is still unconfigured but inserted in a PCI
slot.
The TE120P's jumper is set to E1 as it will connect to
a commercial PBX's PRI card also
Hello Erick,
I believe that if you go for a manual installation of non-AsteriskNOW
components (like Java) they should be excluded from the components that
Conary mantains.
l.
On Mon, 21 May 2007 09:54:52 +0200, Erick Perez [EMAIL PROTECTED] wrote:
I realized that queuemetrics uses
On 5/21/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
My asterisk server was working with a 4-FXO analog
card (TDM400P).
I recently added two digital cards: a TE120P (1 PRI)
and a B410P (4 BRI).
The B410P is still unconfigured but inserted in a PCI
slot.
The TE120P's jumper is set to E1 as it will
On Sat, 2007-05-19 at 12:12 -0400, Steve Totaro wrote:
I have no affiliation with them but if their quotes are accurate then
they provide quite a few options as far as TDM connectivity and realtime
pricing.
If you do not want a phone call from a sales person, give them a BTN
that goes to an
Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088
On 5/17/07, Tim Verscheure [EMAIL PROTECTED] wrote:
Still nothing. I'll give my config files:
manager.conf
;
; Asterisk Call Management support
;
; By default asterisk will listen on localhost only.
--- David Gomillion [EMAIL PROTECTED] wrote:
On 5/21/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
My asterisk server was working with a 4-FXO analog
card (TDM400P).
I recently added two digital cards: a TE120P (1
PRI)
and a B410P (4 BRI).
The B410P is still unconfigured but
yes!!
2007/5/21, Guilherme Góes [EMAIL PROTECTED]:
Did you acess the page at port 8088 ? I.e.: http://192.168.0.1:8088
On 5/17/07, Tim Verscheure [EMAIL PROTECTED] wrote:
Still nothing. I'll give my config files:
manager.conf
;
; Asterisk Call Management support
;
Hi
Ya that works good.
Thanks
Arpit
On 5/20/07, Kapil Dhawan [EMAIL PROTECTED] wrote:
Arpit
Use Auto dial. http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Create a .call file as mentioned by Dave.
Dave Miller wrote:
Arpit Mehta wrote on 5/19/07 10:18 PM:
I was just wondering how
Vieri,
Make sure you are loading the digital card first and then analog card.
I had the same problem and Digium engineers helped me out.
Cheers,
Nitesh
Vieri wrote:
--- David Gomillion [EMAIL PROTECTED] wrote:
On 5/21/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
My asterisk server was
On 5/21/07, Vieri [EMAIL PROTECTED] wrote:
--- David Gomillion [EMAIL PROTECTED] wrote:
On 5/21/07, Vieri [EMAIL PROTECTED] wrote:
Hi,
My asterisk server was working with a 4-FXO analog
card (TDM400P).
I recently added two digital cards: a TE120P (1
PRI)
and a B410P (4 BRI).
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting Explicit MWI Subscription to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Thanks!
--
Warm Regards,
Lee
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I created a *9 extension which executes VoiceMailMain with the callerid
number as the argument. Then of course the voicemail box just has to
be the same as the phone number.
Then we just have another DID for outside access.
*
Adam Moffett
Plexicomm, LLC
[EMAIL
I am running into two problems:
1) The ringing stops during call screening once the extension picks up (but
has not yet approved call)
When a person calls and choose an extension, the Dial link is called and the
person hears the ring -- but as soon as the receiving caller picks up (even
though
Hi folks!
I'm having a problem where my music on hold is just blaring to my
callers. I've tried several different formats (converting using mpg123
and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail.
Every file plays way too loud.
I did notice that sox has a -v flag for
Anthony Francis wrote:
Carlos Chavez wrote:
On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote:
Use the cdr's, who wont know who but at least which phone did it.
I tried following the CDR but if I dial extension 4000 and extension
4002 picks up the call using *8 the CDR says
Cool, please send me the pattern so i can add it
Cheers
Rick
On 5/21/07, Alexandre VERNIOL [EMAIL PROTECTED] wrote:
Really Great!!! Works for me in France I have just change the pattern
and that's ok reallygood job!
Cheers,
Alex
Richard Hamnett a écrit :
Hi there,
Just to announce that
Mike Hammett wrote:
I was looking at the ILECs’ web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during the greeting to
enter the voicemail system? If they call their
Thanks Nitesh,
I did just that and got both the TE120P PRI and the
analog card working together.
The 4-BRI mISDN B410P is a bit tougher and I still
haven't understood yet where the channels are supposed
to be specified (if so) in /etc/asterisk/misdn.conf.
I've used www.misdn.org , NOT the
You could schedule a cron job to run a shell script to delete any files in
the //voicemail/*/Old/ directory that are older than the amount of time
specified. You could craft something up by comparing the date modification
timestamp from `ls -l` or the access modification from `ls -lu`(?). I
On Mon, 2007-05-21 at 14:39 -0400, Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting Explicit MWI Subscription to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Thanks!
In my experience it is never
Now I get this... If I call from 5011 on the 192.168.1.103 machine to
6010 on the 192.168.1.69 machine my X-lite softphone says, call
declined
this is the output:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508,
ext-local|6010|1) in new stack
-- Goto (ext-local,6010,1)
[May 21
Jay Moore wrote:
Hi folks!
I did notice that sox has a -v flag for adjusting volume, but danged
if I can find documentation online that'll tell me what parameter to
pass.
Doing a 'man sox' does wonders:
-v volume Change amplitude (floating point); less than 1.0 decreases,
Hi,
Im new, but trying real hard! I just need general direction, not
details yet..i'll try to figure those...just looking to avoid brick
walls...bottlenecks...inefficiencies etc upfront.
Hardware:
motorola vt2442 - trixbox
Apps:
Dot Net application that operates the Manager API and the FASTAGI
Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting Explicit MWI Subscription to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Did you set the mailbox= variable in sip.conf? I made that mistake
yesterday and
Doug,
Thanks for the reply. Immediately after hitting send I found exactly
what I was looking for. Don't know why I didn't consider doing a 'man
sox' earlier. I must be getting senile. ;)
That said, I altered my initial .gsm files and made them 75% quieter (-v
.25). I replaced my loud
Malcom Kemp wrote:
make[1]: g++: Command not found
hint :)
--
Remco Post
I didn't write all this code, and I can't even pretend that all of it
makes sense. -- Glen Hattrup
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Tim Verscheure wrote:
Now I get this... If I call from 5011 on the 192.168.1.103 machine to
6010 on the 192.168.1.69 machine my X-lite softphone says, call
declined
this is the output:
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/5011-081da508,
ext-local|6010|1) in new stack
-- Goto
Hi,
is there a way or feature available in Asterisk where one can 'pull' a
call back from
voicemail.
i.e. if you don't get to the phone in time and it goes to voicemail,
can you key some
sequence in and pull the caller out of voicemail and speak to them?
Thanks
Mike
Mike Dent wrote:
Hi,
is there a way or feature available in Asterisk where one can 'pull' a
call back from
voicemail.
i.e. if you don't get to the phone in time and it goes to voicemail,
can you key some
sequence in and pull the caller out of voicemail and speak to them?
It seems like you
Anyone have Windows Media streaming for MOH working? I followed the various
procedures on the Asterisk Wiki for using mplayer which seems to be the only
Linux player capable of playing windows media streaming audio (asf, wmv
etc.). Anyone get this working?
I can get shoutcast streams working
Stephen Bosch wrote:
Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in
the web interface by setting Explicit MWI Subscription to true, but
no lights, no stutter tone.
Firmware: 1.4.0.1048
Did you set the mailbox= variable in sip.conf? I made that
shadowym wrote:
Anyone have Windows Media streaming for MOH working? I followed the various
procedures on the Asterisk Wiki for using mplayer which seems to be the only
Linux player capable of playing windows media streaming audio (asf, wmv
etc.). Anyone get this working?
I can get
Stephen Bosch wrote:
Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in
the web interface by setting Explicit MWI Subscription to true, but
no lights, no stutter tone.
Firmware: 1.4.0.1048
Did you set the mailbox= variable in sip.conf? I made that
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in
the web interface by setting Explicit MWI Subscription to true,
but no lights, no stutter tone.
Firmware: 1.4.0.1048
Did you set the mailbox= variable in
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Jay Moore wrote:
Hi folks!
I'm having a problem where my music on hold is just blaring to my
callers. I've tried several different formats (converting using mpg123
and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail.
Every
This is probably cold comfort but I have NEVER had any issues with MWI
working on Aastra phones. It always just works by default. No extra
configuration necessary on the phone for sure. Just reset it to factory
defaults. Explicit MWI is NOT checked by default and I have never had to
check it.
Henry wrote:
Hi,
Im new, but trying real hard! I just need general direction, not
details yet..i'll try to figure those...just looking to avoid brick
walls...bottlenecks...inefficiencies etc upfront.
Hardware:
motorola vt2442 - trixbox
Apps:
Dot Net
make[1]: g++: Command not found
You have just to install cpp
Alex,
Malcom Kemp a écrit :
Thanks to all that have helped me so far. I have made a lot of
progress. I am able to make prilib and zaptel. Now to Asterisk…
After installing the kernel source, I have:
# cd /usr/src/linux
# make
Anthony Francis wrote:
Lee Jenkins wrote:
Stephen Bosch wrote:
Lee Jenkins wrote:
I need to setup MWI on a few Aastra 9112's. I've tried doing so in
the web interface by setting Explicit MWI Subscription to true,
but no lights, no stutter tone.
Firmware: 1.4.0.1048
Did you set the
shadowym wrote:
This is probably cold comfort but I have NEVER had any issues with MWI
working on Aastra phones. It always just works by default. No extra
configuration necessary on the phone for sure. Just reset it to factory
defaults. Explicit MWI is NOT checked by default and I have never
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote:
Mike Hammett wrote:
I was looking at the ILECs? web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during
hi,
I am trying to check my voice mail on a new asterisk instalation.
I get the standard voicemail menu, but when I press any button , it
does not accept the option. It keeps repeating the menu and then
exits ...
Any suggestions ?
thanks and regards
Anand
--
If it is easy, could you enlighten me? I have another thread on caller ID
matching, but I haven't received any positive responses.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Sounds like wrong type of dtmf signaling if you are using a SIP phone.
Check the settings fordtmfmode in sip.conf.
dtmfmode= rfc2833 ; Choices are inband, rfc2833, or info
your choice might not have been right ;-)
If your phones uses ZAP try
relaxdtmf=yes
Frank
If I had to make a wild guess, I'd expect that when you make a call
off-campus you must dial an access code first.
Looking at columbia.edu, I see that you're expected to dial '93' for a local
number.
1+ is a number in the Centrex dial plan for the Morningside campus.
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