On Mon, Aug 23, 2010 at 5:06 PM, Infra m...@waste.org wrote:
On Aug 7, 2007 'Mojo' wrote:
Nicholas Blasgen wrote:
I've got 4 SIP phone lines with a call-limit of 2 for each. I've
written a handy macro to allow my users to dial a phone number and the
macro will figure out the next available
On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be
wrote:
intro extended version.wav: RIFF (little-endian) data, WAVE audio,
Microsoft
PCM, 16 bit, stereo 44100 Hz
You need *MONO, 8000Hz*
$ man sox
--
Motiejus Jakštys
, Ringing, Hangup events and store that
info to database with very accurate timestamps :-)
http://github.com/Motiejus/Asterisk-perl-AMI/blob/master/asterisk_ami.pl
Regards,
Motiejus Jakštys
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(Priv: system,reporting,all)
From:
http://www.voip-info.org/wiki/view/Asterisk+manager+API
And maybe this (check output):
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+SIPshowPeer
Regards
Motiejus Jakštys
, but I have simply not been able to turn this up.
Philipp
P.S.: This is all about audio analysis, not about cause codes.
Exact match:
http://github.com/Motiejus/SoundPatty
Regards
Motiejus Jakštys
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On Wed, Aug 4, 2010 at 12:12 PM, Janu Mukherjee janu.mu...@gmail.com
wrote:
Hi,
Hi, please learn to ask questions.
I have an xlite registered with asterisk server. When i dial a number AGI
is
invoked. and in this we are running *to threads one to record files and
one
to play files.*
What
On Thu, Jul 29, 2010 at 2:32 PM, Benny Amorsen benny+use...@amorsen.dk wrote:
Sherwood McGowan sherwood.mcgo...@gmail.com writes:
I'm going to go ahead and say that while I'm not one of the
developers, I think it's safe to say that you cannot record to a file
and play it back at the same
On Sun, Jul 25, 2010 at 3:11 AM, Norbert Zawodsky norb...@zawodsky.at wrote:
Hello again!
after it being relatively quiet her for the last weeks, my Astrerisk
server was the target of 3 of that nasty REGISTER attacks during the
last days. While I can see not much danger coming from these
configuration - it will.
Regards
Motiejus Jakštys
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http
Hello, Asterisk party,
If block the call before dialing (Hangup()), CDR's don't write to
MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write
normally.
Here is the dialplan:
; we skipped dial, because the number is blocked
exten = _X.,n(Finish),Hangup()
exten = h,1,NoOP(hangup)
exten
If you can install python or PHP in that machine (in means of
storage), you are free to run it there. 64 RAM is really enough to run
python, so you have to just try if it suits in the application. If it
takes too slow to initialize - try to find some embedded versions.
openwrt, for instance, has
Check CPU usage. Maybe one asterisk throttles it for all?
On Sat, Jun 19, 2010 at 12:21 PM, michel freiha mich...@gmail.com wrote:
Dear All,
I have installed 4 asterisks on the same Centos machine..Each Asterisk has
its own installation folder and use its own libraries...Everything looks
Julien,
Just for the record, you don't need registration to iptel.org - just
plain DIAL(SIP/iptel/music).
On Sun, Jun 6, 2010 at 11:47 PM, Julien Claassen jul...@c-lab.de wrote:
Thanks anyway, Ira. It was very kind of you to help me along as far as you
could. I appreciate it.
anyone else
On Fri, Jun 4, 2010 at 11:52 AM, Raimund Sacherer r...@runsolutions.com wrote:
Hello,
We have a scenario in which there are 2 sites, one in europe and one in
mexico, they are connected via an IAX channel, problem is that the location
in mexico has only a dynamic IP connection to the
Just googled it:
http://www.iptel.org/service
Echo test call
Call echo (sip:e...@iptel.org) or the vanity number 3246 for an echo
test call. You can change the buffering while in the call by pressing
the star key.
Music test call
Call music (sip:mu...@iptel.org) to listen to a wonderful fado of
Hi Julien,
I remember I had a similar issue with Jack_hook trashed sound. If I
remember well, caller heared me well, but I heard garbage (or
vice-versa, I don't remember accurately). Sorry, I don't remember how
I solved the problem (it just disapeared after changing something).
However I can
Just needed to install libresample1 and libresample1-dev and all worked great.
Thanks for suggestions.
Because of the (2) reason I am planning to execute this line:
*CLI core set chanvar SIP/$channel
JACK_HOOK(manipulate,i(SIP/$channel:input),o(SIP/$channel:output)) on
Through AMI. Executing
Hi,
I have been headbanging with asterisk and Jack for a while, decited to
ask other linuxists for an advice.
The problem is that Jack is compiled from source (0.118) in /usr/local/, but
menuselect says XXX for it (cannot enable it). I need jack...
Otherwise I will inotify Monitor WAVs, what is
/.app_jack.makeopts
-rw-r--r-- 1 motiejus motiejus 166 2010-05-26 15:12
./apps/.app_jack.moduleinfo
motie...@pbx3:/usr/src/asterisk-1.6.2.7$
Any more suggestions?
On Wed, May 26, 2010 at 2:21 PM, cov...@ccs.covici.com wrote:
Motiejus Jakštys desired@gmail.com wrote:
Hi,
I have been
Opened a bug report for it:
https://issues.asterisk.org/view.php?id=17402
2010/5/26 Motiejus Jakštys desired@gmail.com:
Tried the following, both did not work:
jack:
./configure --prefix=/usr make sudo make install
./configure --disable-xmldoc make menuselect - same problem (XXX
Assume previous IP is LAN. Forwarding public IP ports to LAN is
straighforward. However, with SIP headers you will (don't know H323)
have to modify outgoing SIP headers: replace LAN ip with WAN ip.
For callers you have to substitute RTP destination IP
For callees you have to substitute RTP source
Hi List,
Our company has several small distributed offices we would like to
inter-connect with bridged VPN a single subnet (last example in
http://www.shorewall.net/OPENVPN.html). We have SIP phones in every
office (up to 5) so we can use SIP without any NATing and securely.
Max theoretical
On Wed, May 26, 2010 at 8:01 PM, Andrew Hakman andrew.hak...@gmail.com wrote:
I use openvpn for VOIP traffic all the time. It's not a commercial
application, and only one simultaneous call usually on each vpn link,
but I even have a VPN client on a Linksys WRT-54g wireless router with
1 phone
On Wed, May 26, 2010 at 7:28 PM, Julien Claassen jul...@c-lab.de wrote:
Hi Motiejus!
If all else fails for the moment, it should be quite simple to move JACK.
Move all jack applications from /usr/local/bin to /usr/bin.
In /usr/local/lib move the dir jack and libjack* to /usr/lib.
That
Hi Julien,
yes, I am thinking about implementing something better than asterisk,
but it is a future talk. It really lacks support for flexibility :S
I would be grateful if you found the hacked configure script and
sent it to me :-)
I did not really understand the part about C program, could you
I am testing Jack in asterisk 1.6.2.7 now, using JACK_HOOK and channel
variables (connecting parties to jack during the call).
It works, but it chokes. Every 2-3 days hangs my asterisk. Debugging
it now, trying to find the reason.
So far could run only the standard Debian repo jack (0.109.2,
If the version either 1.4.x or 1.6.x, run make menuconfig from
asterisk source directory, pick up sounds you need, and make install
then.
Or, you can do as mentioned above: install sounds you need explicitly.
On Mon, May 24, 2010 at 5:42 AM, ayodele abejide
ayodeleabej...@hotmail.com wrote:
hi,
Hi, List,
I am looking for a cheapest (and therefore most funny) way to attach
GSM card to my asterisk home box.
Needed features:
Calls+SMS in/out
one or two SIM cards (ports)
Should I try looking for a GSM PCI card that is compatible with
linux/asterisk, or GSM USB card, or modern full-blown SIP
I use Jack for getting callee sound. Dial with option M():
JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on
This works fine, but I need to connect the sound channel to Jack
*before* the actual answer.
As you can see in the AMI log, between Ringing to JACK_HOOK
try with gsm format , what you say?
regards
Dhaval
2010/5/18 Motiejus Jakštys desired@gmail.com
Hi,
The record is not double faster, it's 50% faster (100 seconds original
record - 66.6 seconds recording). Reducing tempo by 33% without
losing pitch sort of fixes the situation, although
Please check WAV headers, what is the sample rate of the file? It
should be 8kHz. Does the WAV sound normal when you decrease sample
rate by hand?
You can just upload one WAV for testing - I'll say what may be wrong with it.
On Tue, May 18, 2010 at 9:52 AM, DHAVAL INDRODIYA
2010/5/18 Motiejus Jakštys desired@gmail.com:
Please check WAV headers, what is the sample rate of the file? It
should be 8kHz. Does the WAV sound normal when you decrease sample
rate by hand?
You can just upload one WAV for testing - I'll say what may be wrong with it.
On Tue, May 18
I am announcing sound recognition project/library SoundPatty. It is
created to capture a recording in an audio stream. Use cases:
You can listen to live radio station and log how many advertisements
are played per day
You can know if leg B is an amazon.com bot :-)
You can match special operator
Talking about file permissions, on Linux everything is possible using
POSIX ACLs. You can set specific rights to files/directories for
certain users.
Note 1: if setting group permissions is enough, use that.
Note 2: Asterisk and web server should be on separate machines (at
least virtual machines)
Hello,
I need to store some additional CDR data from the dialplan, like in
example here (down of the page):
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
However, neither CSV, nor MySQL CDRs have any of these values as the result.
Can you please highlight where can I find the
Issuing HTTP request from dialplan is simple: Use System call when you
have all the statuses:
exten = _X.,n,System(curl -d number=${EXTEN},status=${STATUS}
http://mywebsite/)
Check your dialplan when you have to issue the command
and
man 1 curl
Good luck
On Tue, May 11, 2010 at 12:21 PM, Zhang
If I understand well - you want second PBX to act as your sip.provider.com
add this to your /etc/hosts (on primary pbx):
10.10.10.10 sip.provider.com
(secondary - simulation pbx):
127.0.0.1 localhost sip.provider.com
And use primary pbx as normal. When you need to switch to production -
remove
Forgot to mention that 10.10.10.10 must be the address of your secondary pbx
:)
2010/5/10 Motiejus Jakštys desired@gmail.com
If I understand well - you want second PBX to act as your sip.provider.com
add this to your /etc/hosts (on primary pbx):
10.10.10.10 sip.provider.com
(secondary
-ab23jadf234:input),o(SIP/poly1-ab23jadf234:output))
on
Then all works fine and you get leg B's channel.
-- Forwarded message --
From: Motiejus Jakštys desired@gmail.com
Date: 2010/5/5
Subject: Re: Getting calee audio in Asterisk (real time)
To: Asterisk Users Mailing List - Non
Dear List,
My Dial command:
exten = _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1))
exten = h,1,
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL}) ; Leg B
The problem is: after answering, [connect-jack] both priorities are
executed, and right
Hi,
Great! I thought I won't see leg B channel while using M(), but I do!
:) M() did my day.
Thanks.
On Thu, May 6, 2010 at 4:29 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
[connect-jack]
exten = _X.,1,NoOp(${CHANNEL}) ; Leg A
exten = _X.,2,NoOp(${CHANNEL})
to jack - current Asterisk application
Outgoing call audio - current Asterisk application
Any idea how I could accomplish this?
Regards
Motiejus Jakštys
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suggestions... Maybe I can do this in totally different
approach?
Regards
Motiejus Jakštys
http://m.jakstys.lt/
2010/5/5 Motiejus Jakštys desired@gmail.com
Hello,
I need to capture calee's audio in real-time in order to capture operator
messages (I've written sound recognition software
in the 1st post.
Good luck
On Fri, Apr 30, 2010 at 5:54 PM, Alexandre Vézina avez...@vencomm.ca wrote:
2010/4/30 Motiejus Jakštys desired@gmail.com
Hi,
please always add asterisk version to your query.
I am using Asterisk 1.4.17~dfsg-2ubuntu1.1 on an Ubuntu 8.04.4 server.
I managed
You can call an external script and call CURL from there (either use
AGI, or Asterisk cmd System).
It depends on your task what to use (perl/bash/C...)
On Mon, May 3, 2010 at 7:47 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Dear All,
Last Week i tried and goggling more on how to call
I am 99% sure you will be able to catch this information in AMI. I
didn't try with call diverts, but it says really alot.
On Mon, May 3, 2010 at 4:41 PM, Dan Journo d...@keshercommunications.com
wrote:
Hi,
I am diverting an incoming call to a mobile phone and a landline using the
Hi,
please always add asterisk version to your query.
I managed to run internet radio (that streams MP3) within asterisk.
Minor change is nescesarry to make it work with random MP3s.
My Dialplan:
exten = _X.,n,Answer()
exten = _X.,n,MP3Player(http://stream.m-1.fm/m1/mp3)
$ cat /usr/bin/mpg123
Here is a starting point:
http://www.voip-info.org/wiki/view/Asterisk+dimensioning
Not really what you need, but still. When you figure out something -
add here :-)
Has anyone put together a public list/wiki/info sheet on what the
various maximums/rules of thumb are? Seems a better idea than
GotoIf($[${CALLERID}:.*333.*]?your_extension) (untested)
Something like that (fix variable name to suitable). Check Asterisk regular
expressions.
http://www.voip-info.org/wiki/view/Asterisk+Expressions#Regularexpressions
On Wed, Apr 28, 2010 at 3:49 PM, wassim darwich wassimdarwi...@yahoo.com
Yers. You have 2.5 options:
Monitor, MixMonitor, (these make 1,5) and JACK_HOOK
On Tue, Apr 27, 2010 at 5:53 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello list,
can a conversation be recorded without the caller or callee having to press
some combination that is defined in
AMI writes event Ringing..., you can catch it and (via the same AMI)
send a soft hangup request.
On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Thank you Zhang Shukun,
I was wondering if it is possible to make one or ring and then stop the
call. But i don't
I did use it for my first asterisk installation, but I moved to Debian
due to things I disliked in arch as a server distro.
I did not and still do not use Dahdi at all
(http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge).
I compiled everything from source without PKGBUILDS, since they
Hi,
On Sun, Apr 25, 2010 at 5:06 AM, mike mosier trixbo...@gmail.com wrote:
Howdy all
1. does anyone know a good voip / sip / qos monitoring tool?
Wireshark is quite good at it
http://wiki.wireshark.org/VoIP_calls
However I could only find it good for debugging, not monitoring
(tcpdump the
Use simple PHP's telnet classes for AMI.
If you need special security - use Stunnel (SSL tunnel) and iptables
on asterisk side for IP forwarding.
This all is really straight-forward, I doubt you need a tutorial
here.. Both stunnel and PHP Telnet have tutorials on how to accomplish
this. You just
Hi,
currently I am writing a sound recognition software that will suit
here pretty well - it can recognize your cell phone's our of radio
coverage or similar operator message. It's GPL, link here:
http://github.com/Motiejus/SoundPatty
Now the program can say if 2 WAV files match (tested with out
I opened a ticket about this:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17217
Remove -c on the init script of asterisk, line 85. Should help.
I was trying it with a xen guest.
On Fri, Apr 23, 2010 at 6:52 AM, Kelvin Chan
kelvin.c...@positronics.com wrote:
And I've just done
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