Displaying 20 results from an estimated 1000 matches similar to: "change destination on digit"
2011 Mar 28
8
asterisk and fail2ban
Is anyone using asterisk with fail2ban? I have it working except it takes
way more break-in attempts than what is set in "maxretry" in jail.conf
For example, I get an email saying:
"The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts
against ASTERISK."
when "maxretry = 5" in jail.conf
Perhaps someone else is experiencing this or has resolved it,
2011 Apr 08
9
send voicemail to multiple emails
Is there a way for asterisk's voicemail to send an email (including
voicemail attachment) to multiple email addresses?
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2011 Jun 21
4
call paging interrupts call when using Mitel 5224
Is anybody using Mitel phones? It appears that when you page a Mitel phone
using asterisk's MeetMe, the paged phone will hang up the call its on to
take the page. Thanks in advance.
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2011 May 27
2
disable sip registration
Is there a way to disable all SIP registration and block any requests? The
reason I'm asking is this particular Asterisk server will just be
originating calls. I've noticed sip attacks where the attacker attempts to
register a user 100x per second causing CPU to rise significantly.
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2012 Apr 03
5
process_sdp: Multiple audio streams are not supported
Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent
Hylafax server using softmodems:
Noticed this in the Asterisk log when trying to send a fax from
Hylafax to Asterisk:
[Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp:
Multiple audio streams are not supported
I've googled a few asterisk tickets that may suggest that yes,
multiple audio streams are not
2011 Apr 05
4
agi voicemail callback
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.
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2006 Oct 10
10
Voicemail Press '0'
Crikey. I can't get this to work!
Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says
"the context for the voicemail box that we're looking for in the dialplan for the jump to the
2010 Mar 29
5
Continue a dialplan when the client hang up the call
Hi all,
When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately.
At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc.
Thanks in advance,
Giang
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2011 May 18
3
asterisk's zombie processes
I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too
many zombie processes. I eventually had to disable the notification for the
alert but why does Asterisk create so many zombie processes, I've see more
than 30 at times and it generally stays in the 20s... just seems unusual and
wondering if it's harmful, thanks in advance.
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2010 Mar 10
1
Diaplan reload command not working
I am a complete newbie, completed editing the extensions.conf file, having problem reloading my diaplan via asterisk console, tried to reload it with diaplan reload command, but it says command does not exist.
Please help
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
2011 Apr 06
2
voicemail call back loop
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
VoiceMailMain. Problem is when user hangs up from checking their messages,
it runs the
2006 Feb 01
1
Digit timeouts vs includes in diaplan
Hi,
I have a little situation with my dialplan, and I am wondering if what I
want is even possible.
Here it is: I have three contexts, context1 includes contexts2, and context2
includes context3. In other words, in context1 all extensions of context2
and context3 are valid (and actually working, so that's good). I am using
those context for the sake of code clarity and reuse, and for
2008 Jun 11
2
Losing CDR(accountcode)
Hi,
I`m occassionally seeing CDR(accountcode)'s value empty at a place in my
diaplan where it was filled with some value a few lines before, with nothing
else having changed it.
It`s giving me headaches (as I rely on it for MySQL queries). Anything I
can do?
Mick
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2011 May 16
2
AMI perl daemon
Would anybody know how to run a perl script as a daemon that would stay
connected to asterisk via AMI?
Right now, my AMI script connects to the manager interface, originates a
call, disconnects. The script will be run maybe 20+ per minute. It would
make more sense to me to have the script run as a daemon and have
a persistent connection to asterisk's AMI. Thank you in advance for your
input.
2011 Feb 23
2
REFER and dialplan broken (as documented in chan_sip.c on line 11951)
There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.
Here is issue as stated in chan_sip.c
"this is currently broken as we have no way of telling the
2004 Sep 01
1
Dynamic dialplan
We intend to use Asterisk with a very large dialplan (with a lot of
functionality for 3000+ users). Each user will be able to change several of
his parameters in the dialplan, so we will be forced to reload the diaplan
constantly. Has anybody else any previous experience with a similar
installation? There are some things that we'd like to know, if anybody can
help us. These are:
- Is
2011 Apr 27
2
asterisk practices
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
(using mysql odbc) or terminate with wrong number message
if a message is left in a
2009 Apr 14
2
Exit Dial Application
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
I' try to implement an automatic callback mechanism, just for local SIP calls.. Callback
on busy and on no answer. If the other party doen't answer, it should be possible to press
5 to place an callback.
Here is my dial:
exten => _X.,1,Set(EXITCONTEXT=callback)
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
And here the script for
2016 Jun 30
4
how to join 2 channels using AGI/AMI
Dear all
i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is
possible to configure a scenario like this:
1) receive a call and put it on-hold in a queue (OK)
2) monitor the queue and trigger an outbound call to a remote number using
AMI, setting the channel of the on-hold on a specific var named
channel2Link (OK)
3) when the remote number answer, trigger an
2011 Mar 23
7
asking for some help
hi evrey one,
i'm in some kind interesting in developping some asterisk programme like
doing a small programme including some of these services that do a telephone
operator.
but abviously i need to know about programming in asterisk in thos to files
i think :) (extensions.conf and in sip.conf files)
so i'm asking if someone can give me a puch,i will be very glad
thanks in advance