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
Lately there have been a number of messages about urgent issues. This
is intended as a reply to them all, not just to this specific one.
On Tue, Jul 27, 2010 at 11:08:24AM +0530, Janu Mukherjee wrote:
Hi,
[Snip description of a problem]
How can i achieve this???Please help me in this regard
Hi Dan,
I can see the path does exists but i cant see any recordings happening inn
there.
There are no files in it
Following is the output:
/var/lib/asterisk/sounds
drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings
I hope m understandly this correctly but m sure m missing
Hi All (reposting after 24 hours).
I will do a test call from a soft phone to my mobile. I can speak into my
headset and the audio is heard instantly. But if I speak into my mobile
there is a 1-2 second delay in the Audio. I am using SIP.
I am only finding it in the Zoiper Softphones that we are
Blocking SIP traffic is still going to break ENUM.
The problem with your suggestion Norbert is that Asterisk still would have to
process the requests at an application layer, providing no real advantage to
users of boxes with no grunt.
You could potentially write something to do inspection
Hi Danny,
Thanks a lot for the reply , I tried the dial plan you have provided ,
when pressing '0' it gets connected with the extension defined in
[meetme-oper] context , but after disconnecting the operator call user
exits from the meetme room , can we avoid that ?
I am trying to achieve
Do you see the issue when calling between two softphones? Do you see the issue
if you call from your mobile into an echo test?
Setting TOS flags on packets will make no difference unless the gear in between
is configured to treat them differently. Not that I envision this is the issue
at all.
Hi.
Don't know if you have static public IP, but guess not, so you will have
to configure one dynamic dns service.
There's some services available like www.no-ip.com or www.dyndns.com. I
know that no-ip.com got a linux client, that you can install on your
Asterisk server.
Then you have to
Thats great,
However I need to find a solution to this very problem, not able to code
something from scratch.
Even this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
No, neither that didnt work.
Even this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
I tried this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log
Great, but how exactly do i find that channel - that is my question - which
command.
I am using ruby instead of agi - and i am looking for a command to capture
it in ruby.
I tried this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
Hi!
Great, but how exactly do i find that channel - that is my question -
which command.
For the third time: Use the M option to Dial() and create a Macro. In
that macro use the SIPCHANINFO() or CHANNEL() function to get what you
want to get. No AGI (and AGI is a protocol while Ruby is a
Am 27.07.2010 08:42, schrieb Motiejus Jakštys:
If all you need is block the SIP traffic from external sources, you
may do the following:
# iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT
# iptables -A INPUT -p udp --dport 5060 -j DROP
# iptables-save
Which version of Asterisk are you running?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 2:19 AM, Zarko Zivanovic wrote:
Great, but how exactly do i find that channel - that is my question - which
command.
I am using ruby instead of agi - and i
Hello Motiejus, Hello Nick!
thanks for your answers. My OP was definitely not meant as a request for
help. I just wanted to start some small discussion.
The point is that
a) I don't know fail2ban, and
It's really easy. I just installed it on my company asterisk box - it
took ~5 minutes to
Colin,
I'm working for Zoiper, you can contact us directly on supp...@zoiper.com
Zoa
Nick Brown wrote:
Do you see the issue when calling between two softphones? Do you see the
issue if you call from your mobile into an echo test?
Setting TOS flags on packets will make no difference
Hi, i've some trouble with an * installation when the following scenario
happen.
1) Inbound call to SIP/ ;
2) Call is redirected to ring group 6xx
3) SIP extension 1xx answer.
4) caller want to speak with john doe on his mobile
5) assistant put caller on hold
6) assistant start a call
Hello,
I want to reduce the bandwidth taken by an IAX trunk when used with a small
number of voice channels.
When only one call is passed through an IAX trunk, the IP overhead is indecent.
I would like to increase the IAX voice packet emission interval (20ms - i'm
using speex) to something
On Tue, Jul 27, 2010 at 12:45 AM, Faisal Hanif fai...@vopium.com wrote:
Did any one got it solved? If yes how?
Yes, read doc/backtrace.txt. It will explain how to generate an
unoptimized backtrace, then uploaded it to the mailing list.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber:
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco = 1,self/caller,Playback,tt-monkeys
But it's not working yet.
On Mon, Jul 26, 2010 at 11:33 PM, Tilghman Lesher tles...@digium.comwrote:
On Monday 26 July 2010 15:20:26 Felipe Figueiredo wrote:
To UNSUBSCRIBE or update options visit:
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Hi,
I have xlite client registered with a user. Now i dial an extension say 1500
which
has the dial plan as follows.
exten==1500,1,AGI(localhost//hello.agi)
So when i dial extenstion 1500 the script hello.agi is invoked which in turn
plays a welcome message. I now want to transfer the call now to
Thanks to everyone who replied.
This is great news ;).
I'll get the thing upgraded tonight (when it's quiet).
Thanks again.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: 26 July 2010
The problem sounds like fail2ban is failing to write the new rules to a
permanent file, which would otherwise allow the rules to persist after a
reboot.
Tilghman,
That is exactly right. I'm thinking I need to revise the SuSEfirewall init
scripts to follow up with restarting fail2ban,
On Tue, Jul 27, 2010 at 05:42:01PM +0530, Janu Mukherjee wrote:
Hi,
I have xlite client registered with a user. Now i dial an extension say 1500
which
has the dial plan as follows.
exten==1500,1,AGI(localhost//hello.agi)
Obviously this is not the dialplan you have, as this would fail to
Use dial application along with (agi command) exec
for more see
http://www.voip-info.org/wiki/view/exec
On Tue, Jul 27, 2010 at 10:38 AM, Janu Mukherjee janu.mu...@gmail.comwrote:
Hi,
I have xlite registered with a user. Now i dial an extension say 1500 which
has the dial plan as follows.
We had attempted adding the 'r' to the dial command parameter and that
didn't seem to have an effect. We played around with the progressinband
a bit and tried to see if we could find a solution and only ended up
with same results no matter if it were set to yes, no, or never.
We set everything
El 27/07/10 07:12, Janu Mukherjee escribió:
Hi,
I have xlite client registered with a user. Now i dial an extension
say 1500 which
has the dial plan as follows.
exten==1500,1,AGI(localhost//hello.agi)
So when i dial extenstion 1500 the script hello.agi is invoked which
in turn
plays a
Here's something that should be easy for RUBY pro's.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
- Zarko Zivanovic outlaw...@gmail.com wrote:
Here’s something that should be easy for RUBY pro’s.
Here's something that would be infinitely easier:
You could understand that this list isn't your personal technical support
resource where you can delegate how urgent or not your
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
- Zarko Zivanovic outlaw...@gmail.com wrote:
Here's something that should be easy for RUBY pro's.
My .02
On 07/27/2010 09:38 AM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson
*Subject:* Re: [asterisk-users] Urgent help = RUBY AGI
- Zarko Zivanovic outlaw...@gmail.com wrote:
Here’s
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,
perhaps you will also be this Ruby Pro that you speak of J
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/asterisk-users
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Hi,
I missed the beginning of this thread but you or anyone else looking
for help with Ruby + Asterisk should contact Jason Goecke (@jsgoecke
on Twitter or if you don't do Twitter you can look for contact info
there http://twitter.com/gsgoecke).
Jason probably knows as much about that world as
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On Mon, Jul 26, 2010 at 11:34 PM, bruce bruce bruceb...@gmail.com wrote:
I seem to not be able to find any good open source Asterisk Queue Analyzer
and Asterisk Log Analyzer on the web.
google 'freepbx'
It does some of what you want. For the rest of what you want, strongly
consider paying a
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8
testing process. Please report any
in advance!
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zarko
Zivanovic
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
snip
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that
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thanks, i would try all the options out. I am very grateful
_
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On Monday, July 26, 2010 09:55:38 am Tzafrir Cohen wrote:
I suppose I should make a list of known good packages, and put it on
that FAQ page.
GIMP is useless for FAX. Not only does it get the shape of the images
wrong, it can only display the first page of a FAX. I am not familiar
Hi!
7) if john doe want to speak with caller assistant bridge the two
lines using the transfer function of GXP2000 phone (REFER).
After the transfer in the CDR i can't see the callerid of the caller,
only data of the bridged call is reported.
Any idea on what i can do to keep it ?
On Tue, 27 Jul 2010, Jim Dickenson wrote:
In this email I gave you more detailed information but if you had done
core show application dial on CLI you should have been able to ask
more directed questions.
Maybe RTFHT* should have been the first response :)
*) Read The Frick'n Help Text
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
Maybe RTFHT* should have been the first response :)
*) Read The Frick'n Help Text
Don't know about the Ruby part, but
Here's a strange thing.
I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For
conference rooms we're using Polycom IP6000's. We bought two of them
brand new.
When I configure one phone with a username(SPIDR-3758)/password , it
works fine. The other phone won't register with it's
On 10-07-27 07:56 AM, G star wrote:
How can I change that voice packet rate ?
I think you want to read doc/rtp-packetization.txt in your Asterisk source.
Leif.
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Try to use local channel, and the pass the callerid of the caller to the
local channel, an the later put this in CDR using h extention.
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10
Hello,
I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP
settings in sip.conf in this version also. I am facing a problem when
a SIP client makes a call.
When a SIP client registers to asterisk its status shows 'OK' and it
is able to receive incoming calls. But as soon as
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
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Guys,
I put the option X and used the MEETME_EXIT_CONTEXT and it's working
thanks for the help!!!
=)
On Tue, Jul 27, 2010 at 9:07 AM, Felipe Figueiredo
felipe.figueired...@gmail.com wrote:
1) Ok, I'm using now self/peer on the feature map
2) there's no space in features.conf.
toca_macaco =
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
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Kyle Kienapfel wrote:
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
\\||/
Rod
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:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
not good for Queues at all. It's using the reporting tool from Areski and
Areski has recently released an upgrade to it which again is not what I
want.
There are few other programs that do this but really none that are
There is none for free.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-27 6:12 PM, bruce bruce bruceb...@gmail.com wrote:
:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
not good for Queues at all. It's using the reporting tool from Areski and
Areski has
Try with:
http://queuemetrics.com/download/qloaderd-1.17.tar.gz
http://www.areski.net/asterisk-stat-v2/about.php
http://www.micpc.com/qloganalyzer/
Regards,
This is not the first time on this issue. I post in the past an in
house solution.
On Tue, Jul 27, 2010 at 5:38 PM, bruce bruce
Kyle Kienapfel wrote:
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
I'd think twice about trying this, taking into account the recent
spate of attacks to so many of us coming from Amazon EC2 and
Hello
Le 27/07/2010 20:57, Cassius Smith a écrit :
Here's a strange thing.
I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For
conference rooms we're using Polycom IP6000's. We bought two of them
brand new.
[...]
Any ideas? I'm stumped.
If tour register server is
I need to grab the voicemail WAV file once the voicemail command is done. Is
there a hook to be notified that voicemail is done, and get the name of the
recorded file?
Thanks
MD
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Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan but it feels like I'm reinventing the
wheel.
There's an app_record, and I believe app_dictate
On 7/27/2010 7:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase,
On Tue, Jul 27, 2010 at 4:16 PM, Randy R randulo2...@gmail.com wrote:
Kyle Kienapfel wrote:
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
I'd think twice about trying this, taking into account the
I have a couple of Linksys PAP2T-NA Grandstream HT-502 extensions that are
receiving random DTMF tones on their side, but that are not heard by the
outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have
always had this issue. I am only using SIP on the Asterisk server and all
On 10-07-27 06:08 PM, bruce bruce wrote:
:-) I knew someone would bring up FreePBX. I have FreePBX installed and
it's not good for Queues at all. It's using the reporting tool from
Areski and Areski has recently released an upgrade to it which again is
not what I want.
There are few other
On 10-07-27 08:38 PM, Michelle Dupuis wrote:
I need to grab the voicemail WAV file once the voicemail command is done. Is
there a hook to be notified that voicemail is done, and get the name of the
recorded file?
Look at the 'externnotify' option to voicemail.conf.
Leif Madsen.
--
On 10-07-27 08:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase, rerecord, etc.
I can probably do it through dialplan
That's along the lines of what I was thinking, but how do you trap the DTMF
during record and cause that to end recording? I thought record kept going
until hangup?
MD
From: asterisk-users-boun...@lists.digium.com
How can I found more info about them? The voip-info wiki seems to have a one
line description only
Hopefully I don't have to read the source code to figure out the features;(
Thanks,
MD
From: asterisk-users-boun...@lists.digium.com
The problem is that I need to catch the filename in the dialplan, since I will
be recording several other files and concatenating them with SOX.
MD
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
On Tue, Jul 27, 2010 at 10:05 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Hopefully I don't have to read the source code to figure out the features;(
*CLI core show application Record
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger
Terrific! I tried the same for app_dictate but got a very brief usage
description (but not really a description of the features or what it does).
Is there any other documentation on app_dictate out there?
Thanks
MD
From:
Record does not continue until the end of the call, it records until the
# is pressed or the max duration is reached:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Record
Enjoy
On 7/27/2010 9:00 PM, Michelle Dupuis wrote:
That's along the lines of what I was thinking, but how do you trap the
Hi Felipe,
Glad to know that it worked , could you kindly post the complete dialplan
you wrote to achieve it.
I would also love to test it.
Thanks in Advance
Shiju V.Joseph
From:
Felipe Figueiredo felipe.figueired...@gmail.com
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