On 4/5/2011 4:38 PM, Paul Dugas wrote:
First, this appears to be working for me though I'm not 100% sure of
that and cannot guarantee it will for you in any way, shape or form.
With the lawyering out of the way...
I've seen fail2ban allow more than 500 failed SIP login attempts in
under 30
On 4/5/2011 2:45 PM, Bill Michaelson wrote:
On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote:
Message: 12
Date: Tue, 5 Apr 2011 13:36:21 -0500
From: Sherwood McGowan sherwood.mcgo...@gmail.com
Subject: Re: [asterisk-users] Iptables configuration to handle brute,
Am 05.04.2011 18:50, schrieb vip killa:
I'm wondering if there is a simply way to perform a voicemail callback
feature using AGI.
For instance, a caller leaves a voicemail, the voicemail will then
call the owner of the voicemailbox determined by a database look up.
One possibility: look via
In the last 10 days I've had 4 calls be charged by my provider at
exactly 12 hours longer than the call actually lasted. I think this
as all after upgrading from 1.6.17.1 to 1.6.17.2 but I have no way to
be sure and it's so random. 2 calls 20 minutes apart last week and 2
more today many hours
On Tue, 2011-04-05 at 21:10 -0400, Bryant Zimmerman wrote:
I have deployed several 1.8.3.2 systems as upgrades of customers
systems and now I am seeing random crashes. For some reason the builds
lock up and stop taking sip connections. Existing calls stay on but
when the user hangs up no new
We also see the random freeze of asterisk 1.8.3.2. We do use realtime.
I have just applied the patch and will see how our environment holds.
I will report back to the issue mentioned by Ishfaq
Michel Verbraak
*InterCommIT bv* **
On 06-04-11 09:44, Ishfaq Malik wrote:
On Tue, 2011-04-05 at
On Tue, 5 Apr 2011, Steve Edwards wrote:
On Tue, 5 Apr 2011, Gilles wrote:
I'm no expert of iptables, and it seems like it can handle banning
IP's that are trying to register and fail too many times.
Is there a good iptables configuration that I could use as reference?
Gordon Henderson
hey just change following
[status-one-en]
exten = 100,1,Meetme (12345,qdM)
exten = 100,1,Hangup()
Channel: Local/100@status-one-en
CallerID: Rick 55
MaxRetries: 0
RetryTime: 15
WaitTime: 45
Application: Playback
Data: my_status_message
On Mon, Apr 4, 2011 at 10:38 PM, D. Rick
Hello Everyone;
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I
On 6 Apr 2011, at 11:54, Silver Thorne wrote:
Does anyone know of any opensource or otherwise solutions out there that I
can try out?
Asterisk. Google it. If you're too lazy, Google MixMonitor. If you're too lazy
for that:
http://www.voip-info.org/wiki/view/MixMonitor
S
--
On Wed, Apr 6, 2011 at 5:54 AM, Silver Thorne szilvertho...@gmail.comwrote:
Hello Everyone;
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I have
to modify it to make it easier to use, I do not mind.
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I can try
What about a executing an AGI script with:
[general]
externnotify = /some_agi_script.agi
Would that work?
On Wed, Apr 6, 2011 at 3:20 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 05.04.2011 18:50, schrieb vip killa:
I'm wondering if there is a simply way to perform a voicemail callback
try this!!!
http://www.voip-info.org/wiki/view/Asterisk+tips+callback
On Wed, Apr 6, 2011 at 5:30 PM, vip killa vipki...@gmail.com wrote:
What about a executing an AGI script with:
[general]
externnotify = /some_agi_script.agi
Would that work?
On Wed, Apr 6, 2011 at 3:20 AM, Thorsten
${CALLERID(num):-4}
On Tue, Apr 5, 2011 at 2:53 AM, satish patel satish...@hotmail.com wrote:
Perfect! Thanks
what about :-4 ? I want to remove some digits
-satish
--
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Mon, 4 Apr 2011
On 6/04/2011 4:27 AM, isr...@gmail.com wrote:
Ok thanks I found the problem
Your welcome, can I take it that you captured the packets, you then
viewed them in Wireshark and that is how you discovered the issue?
Larry.
--
On 1/04/2011 13:04, Khaled W. Chehab wrote:
Dears,
I have two questions
1-Is there a way to export fax tiff file image from .pcap captured file .
In other words i am trying to backup all faxes that are passing on my
network,and export the fax file later on.
Is this feasible and
On Wed, 6 Apr 2011, vip killa wrote:
What about a executing an AGI script with:
[general]
externnotify = /some_agi_script.agi
Would that work?
No.
What makes a program (compiled or interpreted script) an AGI is that it
follows the AGI protocol.
Very simplistically, the AGI protocol
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, April 06, 2011 6:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call recording - methodology
I am
Am 05.04.11 20:35, schrieb satish patel:
If i want to watch every phone status Idel or Inuse the how should i use hint
in my dialplan. I meant should i need to specify each and every extension ?
or is there any catch-all extensions ?
-Satish
Hello,
You can use a hint wildcard like
/asterisk/astrec/20110406-093637--0559566768-1302096997.5.gsm|av(0)V(0))
in new
stack
[Apr 6 09:36:37] -- Executing [0559566768@default:3]
Dial(Console/dsp, Dahdi/g0/0559566768||tTo) in new
stack
[Apr 6 09:36:37] WARNING[6433]: channel.c:3443 ast_request: No channel type
registered for
'Dahdi
Here's a snippet to place a call using Asterisk Manager (AMI) and PERL
open (my $man_in, /etc/asterisk/manager.conf) or $man_ok=undef;
if ($man_ok) {
while ($man_in) {
if ($_ =~ /^bindaddr/) {
(undef,$man_addr)
On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote:
Sir,
i am using goautodial server , bri card is showing ok but when i try to call
that showing below ,
This configuration is in doing in dubai , so kindly help me how can connet
the call from this ,
what is my mistake is in this
are you using
Asterisk::AMIhttp://search.cpan.org/~greenbean/Asterisk-AMI/lib/Asterisk/AMI.pm
for
this script?
On Wed, Apr 6, 2011 at 10:04 AM, Danny Nicholas da...@debsinc.com wrote:
Yes – I do it that way because I run the module this is included in on
about 10 different Asterisk servers.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, April 06, 2011 10:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi voicemail callback
are you
I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when
someone is left a voicemail it will call the person's mobile phone and
prompt them with the new message. The perl script simply originates a call
to a persons mobile phone and connects it to their voicemail using
2011/4/5 Danny Nicholas da...@debsinc.com
On my Polycom 501’s I use hints to populate a “buddy” list – I hit the
buddies softkey and can see if my “buddy” is on the line.
Hi,
Sorry to hijack this thread but are your Ringing phones displayed as InUse
ones with your setup ?
My
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, April 06, 2011 11:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk hints
2011/4/5 Danny
On 6 Apr 2011, at 17:46, vip killa wrote:
I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that when
someone is left a voicemail it will call the person's mobile phone and prompt
them with the new message. The perl script simply originates a call to a
persons mobile phone and
What happens if there are more than 1 message and the user does not listen
to all messages though?
On Wed, Apr 6, 2011 at 1:00 PM, Steven Howes steve-li...@geekinter.netwrote:
On 6 Apr 2011, at 17:46, vip killa wrote:
I have externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl so that
when
Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try
to playback something I get the following error:
**[WOOMERA]** HW DTMF supported s1c1-
-- Executing [number@from-pstn:1] Answer(WOOMERA/g1/1-7b29, ) in
new stack
**[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29
-- Executing
You are right i believe,
My Polycom 501 not sending subscription to asterisk.
shirley*CLI sip show subscriptions
Peer User Call ID ExtensionLast state
TypeMailboxExpiry
0 active SIP subscriptions
shirley*CLI
Date: Wed, 6 Apr 2011
We've had several customers report since upgrading them to our new
Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer
works. No significant changes have been made to their SIP
configuration, nor to their ATA configuration.
While not exhaustive, these are the ATAs that
does mailcmd send any variables or data to script? I need a way for script
to identify which mailbox was left a message.
On Wed, Apr 6, 2011 at 3:11 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 5 Apr 2011, Steve Edwards wrote:
Use 'mailcmd' in voicemail.conf.
On Wed, 6 Apr
Look like this issue is still there.
From: satish...@hotmail.com
To: satish...@hotmail.com
Subject: RE: IAS trunk error AES encryption disabled. Install OpenSSL.
Date: Wed, 6 Apr 2011 19:45:06 +
look like this issue is still there
From: satish...@hotmail.com
To:
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, April 06, 2011 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] voicemail call back loop
does
On Wed, Apr 6, 2011 at 2:45 PM, satish patel satish...@hotmail.com wrote:
I am getting this wired error when i am calling IAX trunk. Everything
works! but i want to get rid on these RED WARNING messages.. what is wrong
here ? I have func_aes.so module loaded. also i remove and test but still
Hi,
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
hw
--
_
-- Bandwidth and
Yes, I do have that install.
root@shirley:/var/lib/asterisk/agi-bin# dpkg -l | grep -i openssl
ii openssl 0.9.8k-7ubuntu8.6 Secure
Socket Layer (SSL) binary and related
ii python-openssl 0.10-1Python
wrapper
also i have linssl-dev
root@shirley:/usr/local/src/asterisk/asterisk-1.8.3.2/contrib/scripts# dpkg -l
| grep ssl
ii libssl-dev 0.9.8k-7ubuntu8.6 SSL
development libraries, header files and
ii libssl0.9.8 0.9.8k-7ubuntu8.6
res_crypto module was not loaded :)
Whenever i post question and after few min i got answer myself. Magic Sorry
for bother you..
-S
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 6 Apr 2011 19:59:02 +
Subject: Re: [asterisk-users] IAS trunk error AES
I am using Ruby, per call I have 2-4 agi scripts that execute. Each take
0.02 to 0.08sec
On Mon, Apr 4, 2011 at 3:19 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 01.04.2011 14:27, schrieb Roger Burton West:
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote:
Can
Does anyone have any hardware recommendations for a device to take an
incoming DS3 circuit and give me SIP that I can point to my Asterisk
servers. Currently doing DS3 to Adtran but I want to get away from
having PRI cards in all my Asterisk boxes. From looking around I've
found some people
On 11-04-06 03:53 PM, Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
I suggest using res_odbc, it has better support. Aside from
On 11-04-06 03:53 PM, Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
The tables migrate just fine, but you can update them to
Hey Guy!
I have following dialplan for meetme and i want if someone type wrong meetme
extension it should say invalid extension. But look like following doesn't
work. its just hangup if i type wrong number. how to fix this code..
;Conference rooms/lines:
exten = 7580,1,Goto(ivr-meetme,s,1)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Wednesday, April 06, 2011 4:00 PM
To: asterisk-users
Subject: [asterisk-users] asterisk meetme invalid extension
Hey Guy!
I have following dialplan for
On 4/5/11 6:10 PM, Bryant Zimmerman wrote:
I have deployed several 1.8.3.2 systems as upgrades of customers systems and
now I
am seeing random crashes. For some reason the builds lock up and stop taking sip
connections. Existing calls stay on but when the user hangs up no new calls or
reg
On Wed, 6 Apr 2011, satish patel wrote:
I have following dialplan for meetme and i want if someone type wrong
meetme extension it should say invalid extension. But look like
following doesn't work. its just hangup if i type wrong number. how to
fix this code..
exten =
i did and its not working here is console output. We have 8910-8920 meetme
conf room. below i am dialing 8991 for test invalid and its not working..
Packet timed out after 32000ms with no response
== Using SIP RTP CoS mark 5
-- Executing [7580@from-sip:1] Goto(SIP/7527-0030,
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Wednesday, April 06, 2011 4:47 PM
To: asterisk-users
Subject: Re: [asterisk-users] asterisk meetme invalid extension
i did and its not working here is
On Wed, 2011-04-06 at 13:57 -0700, Jonathan Thurman wrote:
On 11-04-06 03:53 PM, Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other
On 4/5/11 6:10 PM, Bryant Zimmerman wrote:
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I
am seeing random crashes. For some reason the builds lock up and stop
taking sip
connections. Existing calls stay on but when the user hangs up no new
calls or reg
At 12:38 AM 4/6/2011, you wrote:
In the last 10 days I've had 4 calls be charged by my provider at
exactly 12 hours longer than the call actually lasted. I think this
as all after upgrading from 1.6.17.1 to 1.6.17.2 but I have no way
to be sure and it's so random. 2 calls 20 minutes apart last
On Wed, Apr 6, 2011 at 2:59 PM, Hans Witvliet h...@a-domani.nl wrote:
[snip]
I think i have to stick with mysql, as info is coming from other
applications, but perhaps some of the other code can be tweaked.
mysql is nice (lots of tiny programs writen for it), but i'm not
religious attached
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per Michel Verbrask's
recomendation. It appers that it has made quite a difference. I don't have an
PRI
connections as all of our PRI's are connected via SIP gateways. I did run into
serveral
for your network it's optional to receive the fax on your server, you can
pass the received fax to the destination, like a voice call with g711 and no
VAD.
ask if you need more info.
best
On Wed, Apr 6, 2011 at 4:55 PM, Bert Van Kets mail...@vankets.com wrote:
On 1/04/2011 13:04, Khaled W.
fail2ban(opensource) is a good choice for you
best
On Wed, Apr 6, 2011 at 1:16 PM, Gordon Henderson gordon+aster...@drogon.net
wrote:
On Tue, 5 Apr 2011, Steve Edwards wrote:
On Tue, 5 Apr 2011, Gilles wrote:
I'm no expert of iptables, and it seems like it can handle banning
IP's
using the realtime functions for voicemail solve this problem.
you can insert a query from your agi to add new voicemail box.
is it what you need ?
On Tue, Apr 5, 2011 at 10:17 PM, Steve Edwards asterisk@sedwards.comwrote:
On Tue, 5 Apr 2011, vip killa wrote:
Is it possible to create a
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