Displaying 20 results from an estimated 1000 matches similar to: "show calls"
2006 Feb 03
2
can asterisk to say chinese like say english
this is not just playback recorded voice. this is let asterisk say chinese.
how to do this.
there have any ideas?
--
Jeffery
iaxtel Num: 1-700-576-1311
fwdnet Num: 728150
2006 Jan 29
2
Access Codes
Or you can use authenticate() and have it take its 'passwords' form a
text file on your machine.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> trixter aka Bret McDanel
> Sent: Sunday, January 29, 2006 5:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
2006 Jan 23
7
G729a Pass-Through and Recording/Monitoring
Hello,
I am wondering about the ability of a server that is simply passing G729
through it to have the ability to record the calls. I know for
voicemail, meetme, and things like that to work, a G729 license must be
installed on the machine since there is transcoding going on.
Is this also true for recording of calls? Will I require licensing for
each recorded call? Will the server see a
2004 May 31
4
wake-up call
Hi there!
I just try to play with die wake-up function described in
http://www.voip-info.org/wiki-Asterisk+tips+wake-up
Everything looks fine but there seem to be missing some soundfiles like
"wakeup-menu". Where can I get these files in order to make this feature
usable?
Regards
Julian Pawlowski
2006 Feb 03
2
g729 license question
I am wondering how the g729 licenses are done during calls. If I have N
licenses for g729, and N are in use and an additional call comes in that
requests N+1 to be in use, how does asterisk handle that call?
Does it dump it? Does it negotiate another codec automagically?
Basically what happens to that call, obviously it wont (shouldnt) let
you use more licenses than you have available, but
2005 Jul 19
12
Best VoIP provider
It does not look like Nufone is still in business, judging from the
content on their site, which is very little. There is not even a
configuration document to download, to connect to their network.
The rates file is only for US/Canada calling. No international
rates on this rates.csv file.
I have signed up with a $5.00 account with them way back in November
2004. After signup, I havent received
2006 Jan 09
7
Presence support on GrandStream GXP-2000
Hi folks,
Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)?
Cheers,
Richard.
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2006 Jan 07
1
re: where can i find all .C files
hi all,
i m using debian to run my asterisk
gateway.I want to make some customization in voicemail
application.For that i need to modify voicmail's
.C(source file) file. can any body tell me where
exactly all .C files resides in the system..........
thanks
tejas
__________________________________________
Yahoo! DSL ? Something to write home about.
Just $16.99/mo. or less.
2006 Jan 20
5
iDEFISK (mac iax2 softphone) release
]
Hey ho,
A few days ago we released the linux version of the phone, today we are
very happy to have the mac version ready for a little field test.
Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php
At the same time, we also put a newer version of the windows and linux
versions online.
Let us know how you feel about it, a more mac look (brushed metal) is
coming.
2006 Jan 17
2
Is Asterisk the right tool?
I want to create a VoIP solution to allow many members of a closed community
to talk to each other (one on one) via soft phones. In many ways, what I
want is not unlike Skype, except that it would allow for relative anonymity
and be open only to a select group. The system should support as many as 500
simultaneous users located on different continents.
Is Asterisk an appropriate tool to use in
2006 May 31
1
INFO: TFOT book- n priorities and labels
Regarding my earlier post about labels and the 'n' priority:
The TFOT book covers the use of these. See the box on page 81 entitled
"Unnumbered Priorities."
http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
-MC
2006 Jun 02
2
Restricting amount of incoming calls
If i get a 8XX number, my provider told me that they will send all the
calls he gets. But due to bandwidth and asterisk capacitiy in this
particular installation, the system is only able to handle 27 incoming
calls.
How in my dialplan do I regulate, sending a busy signal, when my
system hits 27 incoming calls?
thanks,
--
-------------------------------------------
Erick Perez
2006 Jan 04
5
Grandstream web configuration utility
I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386.
Browsing to each device by IP address, I can get logged in using admin
and I can see the advanced settings, however, if I try to change the
settings and clicking the Change button, it just brings me back to ask
for the password again..
I can't get into the Status page or any of the Account1-4 pages either.
It just keeps
2006 Feb 23
6
username as extension
Is there a way to have extensions automatically created for
registered sip users ?
I did some investigation and found some hope in chan_sip with
relation to the somewhat undocumented usereqphone option but i may be
totally off track.
All i want to be able to do is send a call to number@ip_address where
the number is the username configured on the phone that has
registered with asterisk
2006 Jan 21
1
Compiling app_cepstral.c into Asterisk - failing
I have searched and found a couple examples on how to put the app_cepstral.c
into Asterisk but it isn't working. I obviously am not understanding the
examples that I have found.
"Copy the app_cepstral.c file to your asterisk source tree (apps folder) .
You'll also need to add a lines like these to the Makefile in that
directory:
APPS+=app_cepstral.so"
I added it into
2006 Jan 12
3
Bridging app
Hi All-
I am trying to create a post call survey application. I would like to:
1. ask the caller if they want to take a survey after their call completes
2. If no, just transfer the call
3. if yes,
4. bridge up another extension
5. wait for that extension to hang-up
6. have the system (not the user) transfer the call to different
extension
that administers an IVR based survey.
Anyone
2006 Mar 20
5
Numbered Voicemails even with delete option!
Hello,
Thought people might be interested in this.
I want my voicemails emailed to a person and not stored on my asterisk
server. However, I want them to have a sequential number. I found that
if I set the option delete=1 in my voicemail.conf file for the mailbox,
then the numbering would keep being restarted.
I wrote this shell script to fool Asterisk into numbering my voicemails
sequentially
2006 Jan 03
7
Dialer
Hello All,
I am having trouble finding a specific * piece of software so I thought
I would see If you guys can help me get my terminology clear.
First off let me premise this with "no, this is absolutely not for doing
call marketing".
I need to make my Asterisk box call a group of people and play them a
message.
My company deals with education so we need to do follow ups if students
2006 Jan 22
3
Installing the none commercial intel g729codecs into Asterisk@Home 2.2?
Hang on.... there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it?
Thanks,
Doug.
-----Original Message-----
From: Francesco Peeters (Asterisk) [mailto:francesco@fampeeters.com]
Sent: Sun 1/22/2006 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
2006 Jan 17
3
Phone still rings while on a call
Hi All
I have some grandstream phones registered to my asterisk and all internal, external, voicemail services etc are working very well.
I am not sure that it is a problem more so an annoyance. If someone dials my extension number or external DDI while I am already in a call rather than skipping to the next priority in the dial plan for example voicemail the line continues to ring and while in