Hello List,
I have noticed for the past few weeks that someone from an unknown IP is
trying to make a call to my Asterisk box, below is the sample content of the
log file. Sometimes the calls are being made every seconds.
Is my system being hack by someone?
Oct 12 09:41:47] VERBOSE[3114]
Hello *,
is the rtpip patch still valid for asterisk 1.6 (with some code
changes, obviously)?
https://issues.asterisk.org/view.php?id=8161
Or, in asterisk 1.6 there is an alternative to using it?
This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11
--- chan_sip.c 2010-10-12
Hello,
what does this message mean ?
[Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401'
onto UDP socket destined for public_ip:2049
I find this in my debug log file when core set debug 25.
Is something failing, or is this just informative ?
Kind regards,
Jonas.
--
Hi,
I'm using the applicationmap in features.conf to allow the user to press *1 and
run a macro which records using MixMonitor.
All was fine in our office (behind a nat). But when I took the sip phones to an
end user (also behind a nat), I found that *1 didnt work for outgoing calls.
But for
Tank,
I want create channel bank with ip04 (Blackfin+uClinux+4FXO/FXS+1NIC).
I think than zaptel can do it with TDMoE and DACS+RBS ,...
Please help me.
regards.
On 10/11/10, Gareth Blades list-aster...@skycomuk.com wrote:
Karim Davoodi wrote:
Hello,
I want to create channel bank in this
Hello,
I am using 1.6.2.9-1+b1 asterisk.with cdr_mysql.
Everything seems workging correctly except cdr logs.
It fills up all data when a call established except src and clid
Wht can cause this and where should i check??
--
_
--
Am 12.10.10 07:57, schrieb Malvin Rito:
Hello List,
I have noticed for the past few weeks that someone from an unknown IP is
trying to make a call to my Asterisk box, below is the sample content of the
log file. Sometimes the calls are being made every seconds.
Is my system being hack by
Hello,
I am wondering how I can make asterisk think that the user is registered on
it..Scheme is the following: user registerkamailio (put data in
location table)--asterisk
All users are registered on kamailio and I want to duplicate that info on
asterisk. If a call comes to kamailio
Hello,
Is there a possibility for asterisk to work out REFER messages in the
dialplan? like INVITES. I need it to distinguish forwarded calls from all
the other calls.
--
_
-- Bandwidth and Colocation Provided by
Check the Asterisk option autocreatepeer. See
http://gnuradio.org/redmine/wiki/gnuradio/OpenBTSThe_use_of_autocreatepeer=yes
for an example. This is specially made if registration is done by a
real sip register like Kamilio.
\Erik
On 12 okt 2010, at 16:10, Borin wrote:
Hello,
I am
On Mon, Oct 11, 2010 at 6:14 PM, Daniel Knoll dan...@danielknoll.de wrote:
Hey,
i forgot to ask, how can i get the user number from a caller he is in a
conference, i don't find a variable to us this for the current channel.
Only the command meetme list roomnr shows the usernumber, but i can't
Hi gang,
I have a fun one for you. I'm not getting the quality of
sound I want out of GSM, so I'm trying to make my files into .WAV (.wav)
format. Here is the file output for 5 files:
file *.WAV
cents.WAV:RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000
You have two separate problems here:
(1)
dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit,
mono 8000 Hz
You should have generated this with 16-bit resolution, like all the
others.
(2)
Not sure about the cents - sure it's coming out as 16-bit? Is the file
in the right
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roger Burton
West
Sent: Tuesday, October 12, 2010 3:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] sound file debug
You have two separate
On Tue, Oct 12, 2010 at 10:05 AM, Stefan Schmidt s...@sil.at wrote:
so if you dont know someone in china, it would be a good idea to block
this AND set allowguest=no to prevent this in future.
And firewall your Asterisk box.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
On Tue, Oct 12, 2010 at 8:08 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
[Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto
UDP socket destined for public_ip:2049
Is something failing, or is this just informative ?
No, this is a debug message. Unless you are
On Tue, Oct 12, 2010 at 4:23 PM, Danny Nicholas da...@debsinc.com wrote:
dollars.gsm: data
dollars.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz
Can't be 100% certain on
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