I have been trying to install cdr-stats for a week now, but there is no
documentation worth the try and the amount of errors is huge. CUrrently
stuck running
python manage.py runserver 0.0.0.0:8000
I get
python manage.py runserver 0.0.0.0:8000
Error: No module named dilla
When starting apache,
The answer function on an analog line is accomplished by going off
hook. Unless the line is controlled by an automated device (like
answering machine) someone has to physically take the device off hook
to answer it. The ATA has no way to do it as all it gives is the FXS
signalling.
What exactly
Kevin P. Fleming kpfleming at digium.com writes:
Yes, it was lost during a merge of code into Asterisk trunk after 1.6.2
was branched (so only 1.8.0 and trunk are missing the code). Leif Madsen
entered an issue on Mantis as a blocker for any more 1.8.x releases
until this is resolved, as
Dear all,
The problem here is that as soon as asterisk dialing on fxo lines it
sets channel status as answered although the chennel is getting
ring back tone from
other party.
Anyone can suggest me to solve this issue ?
Thanks ,
Ashik
On Tue, Apr 26, 2011 at 4:28 PM, Jim Dickenson
On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali
beaasteriskg...@gmail.com wrote:
The problem here is that as soon as asterisk dialing on fxo lines it
sets channel status as answered although the chennel is getting
ring back tone from
other party.
Anyone can suggest me to solve this issue ?
The
Thanks for your solution.
Anybody can explain me why asterisk is unable to detect ringback tone
from PSTN telco ? .
Does anybody successed; to make asterisk to detect ring back tone from
PSTN telco ?
Thanks,
Ashik
On Wed, Apr 27, 2011 at 12:44 PM, Gilles codecompl...@free.fr wrote:
On Wed,
Hi
How to know status of Asterisk,Mysql. PRI lines and other services from PHP
scripts ?
Thanks and regards
Virendra Bhati
+91-9172341457
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New to
HI,
I have installed asterisk 1.6.2.18 with konference 1.7, All things are
working fine but when we start taking DTMF then key 3 not get my asterisk.
When we use landline number(dedicated number) than all DTMF is capture and
asterisk work fine. In case of mobile only key 3 don't work. Strange
When dialing is finished on an analog FXO Asterisk considers it answered. The
solution is to use something that is not an analog FXO like PRI or SIP to a
carrier.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Good morning,
I have a digium wctdm24xxp in my asterisk box, i am not able to see
the callerid when the call is incoming from the fxo line, i live in
Brazil, how can i change the signaling from fsk to dtmf?
Thanks.
--
_
--
Hi,
Consider the following situation :
SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000
SIP/asterisk-001dAGI Tx 200 result=48
SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000
SIP/asterisk-001dAGI Tx 200 result=48
SIP/asterisk-001dAGI Rx WAIT FOR DIGIT 3000
On Wed, Apr 27, 2011 at 09:26:24AM -0300, Antonio Modesto wrote:
Good morning,
I have a digium wctdm24xxp in my asterisk box, i am not able to see
the callerid when the call is incoming from the fxo line, i live in
Brazil, how can i change the signaling from fsk to dtmf?
Hello Antonio,
In
Hi:
http://php.net/manual/en/function.system.php
Then, the commands you shoul run:
/usr/sbin/asterisk -rnxpri show spans
/etc/init.d/asterisk status
/etc/init.d/mysql status
.
.
.
.
and so on!!
good luck!
Regards.
Juan.
Linux User #441131
On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati
Hi, you can use the PHPAgi project
http://phpagi.sourceforge.net/
Otherwise, if you want a more high-level approach you can use the MXML
interface, you will communicate with HTTP GET request and obtaing XML
response directly from Asterisk.
Enabling the http manager interface you will get
Hi list,
I've been beating my head for about 3 days on this one. I have
Asterisk 1.4.41 installed using openh323. As long as I'm inside my
firewall, everything is hunky-dory. When I move to server on another
subnet, I'm still able to connect, but no longer have sound. Any good
pointers
Danny Nicholas wrote:
Hi list,
I've been beating my head for about 3 days on this one. I have
Asterisk 1.4.41 installed using openh323. As long as I'm inside my
firewall, everything is hunky-dory. When I move to server on another
subnet, I'm still able to connect, but no longer have
Hi all,
I'm trying to get my head around our Asterisk network configuration.
We've been using it for about 2 years now (home office) and it works
great. Its Asterisk 1.4.2 with SIP through external provider(s).
We have the Asterisk server behind our IPCop firewall, and have a
dedicated IP
I just completed building a feature rich asterisk voicemail system using
perl, php, and mysql.
My only concern is that the system i built will not be able to handle the
call volume needed. Let me start by explaining my setup.
Incoming call - route.agi (perl - mysql lookup) - AGI - voicemailbox
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jose P. Espinal
Sent: Wednesday, April 27, 2011 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] h323 with
Hey,
You could test it? Setup a second server that calls the voicemail on the
first server. Test different call volumes that way. Write a simple
script that calls asterisk manager and originate your 50 calls
asynchronously, see what happens.
If it's a production server, you may want to wait
On Wed, 2011-04-27 at 10:16 -0700, Myles Wakeham wrote:
Well there is one 'optimization' that I need to sort out. There seems
to be some latency between the Asterisk server (and the SIP Phones) and
callers. Depending on the caller's network (ie. POTS, Cell phone, other
Voip, etc.) we
On Wed, Apr 27, 2011 at 1:16 PM, Myles Wakeham my...@techsol.org wrote:
It kinda scares me though. I know that SIP is an attractive attack-vector,
and that there are scripts out there that target SIP devices. I know I
could run Fail2Ban on the server, which is fine (we're doing that anyway
Are there any internal DHCP or DNS services built-in to the Asterisk code?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
No [1]
1. AsteriskNOW does have some of these services as do many
distributions like Zentyal.
On Wed, Apr 27, 2011 at 2:04 PM, Thomas Perron thomas.per...@gmail.com wrote:
Are there any internal DHCP or DNS services built-in to the Asterisk code?
--
--
No.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Perron
Sent: Wednesday, April 27, 2011 2:04 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DHCP / DNS
Are there any internal DHCP
[Danny Nicholas]
Thanks for the information - but this doesn't seem to play well with SUSE.
Any ideas?
If you are open to the possibility of building from source I think I
might have a little white paper based on the scripts (about installing
latest version of H323plus on 1.4.X) by today,
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jose P. Espinal
Sent: Wednesday, April 27, 2011 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] h323 with
Hi All,
Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2
echo cancellation on channel. If i want to use OSLEC then what should i need
to do ? Do i need to recompile dahdi with OSLEC ?
-S
--
Friends,
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in April
2011 - basically now. After that, only security patches would be committed.
This is already a delay from the original plan
Unfortunatelly that doesn't change anything. I got exactly the same error
(Everyone is busy/congested at this time (1:0/0/1) ... ).
I did a dialplan reload before testing of course.
2011/4/25 Camilo Echeverry camiloecheve...@gmail.com
As I see in your iax.conf, IAX Peer belogs to special
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Olle E. Johansson
Sent: Wednesday, April 27, 2011 2:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Discussion:
- Original Message -
Friends,
SNIP
Unfortunately, I think this is way too early. My feeling and
experience is that 1.8 is not ready for production in the environments
I work in - large scale installations. Customers are not planning
migration and all new installs are still 1.4.
On Wed, Apr 27, 2011 at 3:34 PM, Olle E. Johansson o...@edvina.net wrote:
Friends,
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in
April 2011 - basically now. After that, only security
I(me, my opinion, my feelings, my commercial view) am on the side of
dropping support for 1.4 and 1.6. 1.8 had some major issues which are
resolved/being worked on with more energy as older platforms are shut
down. If a large enough security issue showed up, I hope we would all
try to do
Le 27/04/2011 21:34, Olle E. Johansson a écrit :
Friends,
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in April
2011 - basically now. After that, only security patches would be committed.
On Wed, 27 Apr 2011, Olle E. Johansson wrote:
The Digium team wants to go ahead and not support 1.4 any more, I want
to keep 1.4 open for normal bug fixes. What do you think?
I would like to see continued bug and security fixes for 1.4 for some time
yet.
As well as a raft of hosted
On Wednesday 27 April 2011 2:13:07 am C F wrote:
The answer function on an analog line is accomplished by going off
hook. Unless the line is controlled by an automated device (like
answering machine) someone has to physically take the device off hook
to answer it. The ATA has no way to do it
On 04/27/2011 02:06 PM, satish patel wrote:
Which echo cancellation is good between OSLEC and MG2. Dahdi by default use
MG2 echo cancellation on channel. If i want to use OSLEC then what should i
need to do ? Do i need to recompile dahdi with OSLEC ?
Yes, you would need to compile the OSLEC
I agree 100%, it's too early.
There is a lot of businnes out of there based on 1.4 (even still 1.2),
and my feelings is that a lot of people is not going to upgrade the
asterisk version, they are going to stay with 1.4 for a long time yet.
Also i wanna add another little consideration. Voip
- Original Message -
From: Olle E. Johansson o...@edvina.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 27, 2011 3:34:03 PM
Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
Friends,
We
We have a discussion on asterisk-dev about the maintenance of the 1.4 branch.
According to the release plans, support for 1.4 was scheduled to close in
April 2011 - basically now. After that, only security patches would be
committed. This is already a delay from the original plan published
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