hi, all
i want to try realtime function. but after i install the adds-on . i
cant see the realtime modules have been loaded.
modules exist here:
[r...@localhost modules]# ls *mysql*
app_addon_sql_mysql.so cdr_addon_mysql.so res_config_mysql.so
and i can't find the modules
*CLI module show
On Fri, 26 Feb 2010, Alejandro Recarey wrote:
I'm trying to get my asterisk server to reinvite. I have two asterisk
servers with public IP's. My users (behind NAT) register on one server
(I'll call it server 1), and for some calls they are transfered over
to the other server (server 2),
Hi all.
I want to write a diaplan which can make asterisk act as a karaoke serivce.
It mean that A user can call to Asterisk, and while the user singing a song,
the asterisk play a background music.
Is it possible to do that ? please help me.
Thanks in Advance,
Giangnh
--
Hi,
I have a problem with the Asterisk BLB's. When I call to other extension,
the called extension don't still InUse for more of 3 minutes. The call still
active, but only the caller extension is InUse when I type show hints in
the Asterisk console, the called extension is marked as Idle .
My
Jeff Brower wrote:
Jonathan-
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of
samples between the gaps, does it correspond to multiples of RTP
packet payload length (for example, for 8 kHz G711 multiples of
80
- Gordon Henderson gordon+aster...@drogon.net escreveu:
On Thu, 25 Feb 2010, Vinícius Fontes wrote:
Just checked and I'm using the high res timer as well:
Feb 25 17:42:32 voyage vmunix: [ 27.028798] dahdi_dummy: Trying to
load High Resolution Timer
Feb 25 17:42:32 voyage vmunix:
has anyone ever heard or read of an actual case of someone packet sniffing the
tones to get pin#'s?
_
Hotmail: Free, trusted and rich email service.
- --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
we are running Asterisk 1.6.1.14 and have a issue that when we use
followme the call is correctly placed to the mobile phone, the mobile
rings, but when answered we do not hear the normal followme
introduction message. If we press 1 to
On Fri, 26 Feb 2010, Vinícius Fontes wrote:
http://unicorn.drogon.net/configs/config.2.6.30.1.geode
Drop that into .config in a stock 2.6.30.1 kernel off www.kernel.org
and
off you go. That will produce a kernel with no modules in it.
You'll need to re-make dahdi.
Good luck!
Gordon
--
Hi All,
I would like to know if there is a good web operator/softphone for a
little help desk environment (5-10 people).
Apart from the classic features (call, transfer, conference,...), I
need a small integration with the internal trouble ticket system /
crm. For example when a call
Hi All,
I have installed astriesk 6 and am able to make calls using sip x-lite.
Its working as I expected.
Now I want to make call from sipx-lite to PSTN using asterisk.
can any please suggest me which Hardware card that I can buy? and use ( pl give
me all ths list of cards which are
Hello,
Give our zoiper softphones a try, you could achieve this functionality
by sending a url over IAX (Sendurl) or by using the open website on
incoming call. (In which you pass the callerid as a paramter to the
website to open the ticket that matches that one. (You could also ask
for the
Il giorno 26/feb/10, alle ore 14:59, Zoa ha scritto:
Hello,
Give our zoiper softphones a try, you could achieve this functionality
by sending a url over IAX (Sendurl) or by using the open website on
incoming call. (In which you pass the callerid as a paramter to the
website to open the
On Fri, Feb 26, 2010 at 2:35 PM, Aditya Kumar adityakumar...@yahoo.com wrote:
can any please suggest me which Hardware card that I can buy? and use ( pl
give me all ths list of cards which are good.).
Here is a starting list of Asterisk hardware
http://bit.ly/a6yX6h
--
- --[ UxBoD ]-- ux...@splatnix.net wrote:
- --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
we are running Asterisk 1.6.1.14 and have a issue that when we use
followme the call is correctly placed to the mobile phone, the
mobile
rings, but when answered we do not hear the
On Fri, 26 Feb 2010, Aditya Kumar wrote:
Now I want to make call from sipx-lite to PSTN using asterisk. can any
please suggest me which Hardware card that I can buy?
As an alternative, you can get an account with a SIP or IAX termination
provider.
Outgoing minutes (depending where you are
I have an a bunch of SPA941 Linksys phones for users in and out of the
office. When the phones are in the office (and on the same network as
the asterisk server) the WMI goes on when it should and off when it
should. But when the phone is on another network and natted it fails
to do so.
Thanks Steve and Andy
I did see this option.
But I dont want to depend on the service provider.
I want to convert SIP to PSTN using Asterisk and than connect directly to a
pstn network.
here is what I am looking:
sip lite calls PSTN network
call goes from sip lite to Asterisk
I will make the
I am new to Asterisk and have searched all over for an answer to this,
so please don't skewer me too bad if this is a stupid question. I am
currently running 1.6.0.21 on a few test boxes (one i386, one x64), and
have noticed that there haven't been any RPM updates since .21, even
though .25 just
Hi,
I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name)
feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels
logical I need the
parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf
trunkgroups]
[channels]
language=de
context=default
On Friday 26 February 2010 10:24:41 Jay Vocaire wrote:
I am new to Asterisk and have searched all over for an answer to this,
so please don't skewer me too bad if this is a stupid question. I am
currently running 1.6.0.21 on a few test boxes (one i386, one x64), and
have noticed that there
Hi Folks,
I did not get any responses from the list, but I debugged and fixed it
myself.
Just in case anybody runs into this again here is the problem and the
solution.
When there are two interfaces as in shown in my setup below. Most
examples on the internet talk about peers behind firewalls
Un-top-posting...
On Fri, 26 Feb 2010, Aditya Kumar wrote:
Now I want to make call from sipx-lite to PSTN using asterisk. can any
please suggest me which Hardware card that I can buy?
From: Steve Edwards asterisk@sedwards.com
As an alternative, you can get an account with a SIP or
In article 201002261104.15584.tles...@digium.com,
Tilghman Lesher tles...@digium.com wrote:
Also, if you're using CentOS 5, Digium creates RPMs, which you can source
here: http://packages.digium.com/centos/5/current/
Ah, now that is useful to know. Thanks!
Are these RPMs easily built
Sorry, I did not include enough information. I am using the
Asterisk/Digium yum repositories as detailed here:
http://www.asterisk.org/downloads/yum
I believe I have them setup right, as that is how I did my initial
install of Asterisk (and all of the other dependencies).
Right now, when
Michael Leonetti wrote:
I have an a bunch of SPA941 Linksys phones for users in and out of the
office. When the phones are in the office (and on the same network as
the asterisk server) the WMI goes on when it should and off when it
should. But when the phone is on another network and
On Fri, Feb 26, 2010 at 11:24 AM, Jay Vocaire jvoca...@innproc.com wrote:
I am new to Asterisk and have searched all over for an answer to this,
so please don't skewer me too bad if this is a stupid question. I am
currently running 1.6.0.21 on a few test boxes (one i386, one x64), and
have
Thank you steve for a detail info on all cards.and all...
so
my setup will be
sip-xlite talking to Asterisk
Asterisk Box is connected to USBfxo (example).
USBfxo connects to my Phone line.(which is fxo)
so with translations defined I can call, from Analong phone to sip lite
(internally)
If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find on internet
smime.p7s
Description: S/MIME cryptographic signature
--
_
-- Bandwidth and Colocation Provided by
Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE cipr...@carsenie.ro wrote:
If anyoane have a firmware with sip support for a tainet venus 2804 please
give am feedback caz i kan-t find on internet
--
- C F shma...@gmail.com wrote:
Wouldn't some online translator do a better job? Or just plain old
spell checking?
On Fri, Feb 26, 2010 at 1:11 PM, Ciprian ARSENIE cipr...@carsenie.ro
wrote:
If anyoane have a firmware with sip support for a tainet venus 2804
please
give am
http://www.tainet.net/Product/venus.htm
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Friday, February 26, 2010 1:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Friday 26 February 2010 11:42:42 Jay Vocaire wrote:
Sorry, I did not include enough information. I am using the
Asterisk/Digium yum repositories as detailed here:
http://www.asterisk.org/downloads/yum
I believe I have them setup right, as that is how I did my initial
install of Asterisk
On Fri, 26 Feb 2010, Aditya Kumar wrote:
sip-xlite talking to Asterisk Asterisk Box is connected to USBfxo
(example). USBfxo connects to my Phone line.(which is fxo)
so with translations defined I can call, from Analong phone to sip lite
(internally)
Now if I want to make calls to the
Anybody with a firmware with support SIP for a TAINET VENUS 2804 because I
do not find anywhere on the Internet. Thanks in advance
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent:
Try this link
http://www.epygi.com/forum/archive/index.php/f-14.html
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ciprian
ARSENIE
Sent: Friday, February 26, 2010 12:59 PM
To: '--[ UxBoD ]--'; 'Asterisk
THanks Steve,
I understand about FXO, FXS.
I want to have a connection from my Asterisk box to the External ptstn world
via My home phone line.
So, if I use USBfxo, Can I connect it directly to my wall socket?
so set up will be :
Asterix linux Box--usb haves USBfxo.
that USBfxo is connected to
On Fri, 26 Feb 2010, Aditya Kumar wrote:
I understand about FXO, FXS.
I want to have a connection from my Asterisk box to the External ptstn
world via My home phone line. So, if I use USBfxo, Can I connect it
directly to my wall socket?
so set up will be : Asterix linux Box--usb haves
Nothing :( only www.tainet.net have and i need partner account
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 26, 2010 9:06 PM
To: 'Asterisk Users Mailing List -
Thanks Steve for all the info..
now I am in buying mode :-)
On Fri, 26 Feb 2010, Aditya Kumar wrote:
I understand about FXO, FXS.
I want to have a connection from my Asterisk box to
the External ptstn
world via My home phone line. So, if I use USBfxo,
Can I connect it
directly to my
Thanks for researching this for me. If you actually look at the link
you sent me, you will see that the latest is:
asterisk16-core-1.6.0.21-1_centos5.x86_64.rpm 20-Jan-2010 15:45 11M
So, we come back to my original question: is there a reason for the
delay on getting the RPM's out?
Btw- I am
Jonathan-
Jeff Brower wrote:
Jonathan-
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of
samples between the gaps, does it correspond to multiples of RTP
packet payload length (for example, for 8 kHz G711 multiples
The model is Avaya is g650 and S8720. Software version is 3.1.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A.
Stapleton
Sent: Thursday, February 25, 2010 12:46 PM
To: Asterisk Users List
Subject: Re:
What is the easiest method or can someone point me in the direction I need
to look to do remote agent login..
Ie, Caller calls in with a cell or home phone, authenticates himself, select
a queue to be added too, hangs up, and then any calls coming into said queue
would ring their home or cell
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat
of a band-aid to the issue. But in my observations there is one clear
indicator that I am shocked is not used.
When I have done this test - pulling the network cable on a device during a
call - Asterisk actually reports
Jay Vocaire wrote:
Thanks for researching this for me. If you actually look at the link
you sent me, you will see that the latest is:
asterisk16-core-1.6.0.21-1_centos5.x86_64.rpm 20-Jan-2010 15:45 11M
So, we come back to my original question: is there a reason for the
delay on getting
In article 201002261231.51056.tles...@digium.com,
Tilghman Lesher tles...@digium.com wrote:
Aha. The Asterisk packages themselves are in:
http://packages.asterisk.org/centos/5/current/
Aha, and there are the SRPMS too. Thanks!
Tony
--
Tony Mountifield
Work: t...@softins.co.uk -
So I've been testing asterisk under LXC for a few days now and am very
happy with the results. My test server is a 1.8GHz Celeron with 256KB
cache and 512MB RAM and I have 20 containers each running asterisk (and
apache/php,sendmail and a few other minor things)
More for fun than anything
At 07:52 AM 2/26/2010, you wrote:
Outgoing minutes (depending where you are in the world) should cost less
than US$0.02 per minute with no monthly standing charge.
I'd not been paying attention and was recently surprised to find that
my rates have dipped under .01/minute for prepay at $20 every
Hello,
I have been trying to setup asterisk 1.6.2.0 to receive fax. I have
two SIP trunks connected to asterisk. One of them is a VoIP service
provider and the other is an audiocodes gateway connected with pstn
and fax lines. I am able to receive faxes on the DID numbers provided
by the VoIP
26 feb 2010 kl. 22.02 skrev JT:
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat of
a band-aid to the issue. But in my observations there is one clear indicator
that I am shocked is not used.
When I have done this test - pulling the network cable on a device
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