Re: [asterisk-users] Multiple Asterisk Sessions on same machine
Two very simple solution for your problem: 1-Port redirection in iptables. This I have used for a year or plus and it worked fine for me. I have redirected 1000 ports to a single port 5060 in iptables and it worked smooth. 2-There is a script in asterisk source directory to compile portable asterisk. You can compile asterisk as portable and copy compiled asterisk to multiple locations/directories (as many instances you need). Each copy will have its own configuration files where you can play as you like. Regards, Faisal Hanif -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??
Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option packtes asterisk sends its own information,If you see the below option packet i have highlighted the asterisk word in from field and in from field tag how can i changed it Please let me know same as in User Agent. 192.168.207.70:5060 - 192.168.207.177:5065 OPTIONS sip:192.168.207.177 SIP/2.0..Via: SIP/2.0/UDP 192.168.207.70:5060 ;branch=z9hG4bK57e5b165;rport.*.From: asterisk sip:asterisk*@ 192.168.207.70;t ag=as0977f8f5..To: sip:192.168.207.177..Contact: sip:asterisk@192.168.207.70..Call-ID: 272c85316b257dfa168c9d0155089...@192.168.207.70..cseq: 102 OP TIONS..*User-Agent: Asterisk PBX*..Max-Forwards: 70..Date: Wed, 20 Jul 2011 11:58:01 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTI FY, INFO..Supported: replaces..Content-Length: 0 # Regards, Masood Ahmed masoo...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??
On 07/20/2011 05:00 AM, Masood Ahmed wrote: Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option packtes asterisk sends its own information,If you see the below option packet i have highlighted the asterisk word in from field and in from field tag how can i changed it Please let me know same as in User Agent. These are internally generated, so there is no way to modify them without a source-level change. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??
user-agent could be set in sip.conf On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov abalas...@evaristesys.comwrote: On 07/20/2011 05:00 AM, Masood Ahmed wrote: Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option packtes asterisk sends its own information,If you see the below option packet i have highlighted the asterisk word in from field and in from field tag how can i changed it Please let me know same as in User Agent. These are internally generated, so there is no way to modify them without a source-level change. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Macro to Dial a Channel Group using Round-robin
Good morning, I am writing a Asterisk dialplan from scratch (for learning and testing purposes), but i'm having trouble with a algorithm to dial a SIP group using round-robin. I want that asterisk dial the member of the group in a circular way, until the call be answered. For example, i have the group TEST=SIP/1SIP/2SIP/3SIP/4, asterisk would dial SIP/1, if it doesn't answer in a period of time then asterisk would dial SIP/2 and so on. Can somebody help me? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro to Dial a Channel Group using Round-robin
Try using local channel to accomplish that. by example : you have 2 phones in the group and want to dial those phones in the following fashion. Dial phone 1 first after 15 sec if phone1 does not pickup dial phone2 : [group-call] exten = group1,Dial(Local/phone1@group-callLocal/phone2@group-call, 30) exten = phone1,1,Dial(SIP/100,15) same = n,Hangup() exten = phone2,1,Wait(15) same = n,Dial(SIP/101) same = n,Hangup() Hope that will help. asterisk the definitive guide 2011/7/20 Antonio Modesto mode...@isimples.com.br ** Good morning, I am writing a Asterisk dialplan from scratch (for learning and testing purposes), but i'm having trouble with a algorithm to dial a SIP group using round-robin. I want that asterisk dial the member of the group in a circular way, until the call be answered. For example, i have the group TEST=SIP/1SIP/2SIP/3SIP/4, asterisk would dial SIP/1, if it doesn't answer in a period of time then asterisk would dial SIP/2 and so on. Can somebody help me? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- *Adolphe CHER-AIME Network / VoIP Engineer CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3449-4280* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro to Dial a Channel Group using Round-robin
On Wednesday 20 Jul 2011, Antonio Modesto wrote: I am writing a Asterisk dialplan from scratch (for learning and testing purposes), but i'm having trouble with a algorithm to dial a SIP group using round-robin. I want that asterisk dial the member of the group in a circular way, until the call be answered. For example, i have the group TEST=SIP/1SIP/2SIP/3SIP/4, asterisk would dial SIP/1, if it doesn't answer in a period of time then asterisk would dial SIP/2 and so on. Can somebody help me? Start with this and experiment! You may want to set a variable initially to 0, increment it each time around the loop and use a GotoIf() to jump out after too many cycles. [macro-round-robin] exten = s, 1, Dial(SIP/1, 30) ; dial SIP/1 for 30 exten = s, n, Dial(SIP/2, 30) ; dial SIP/2 for 30 exten = s, n, Dial(SIP/3, 30) ; dial SIP/3 for 30 exten = s, n, Dial(SIP/4, 30) ; dial SIP/4 for 30 exten = s, n, Goto(1) ; start again -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Macro to Dial a Channel Group using Round-robin
Oh man, so easy, thank you very much! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: How can I Add my own Word in option packets in from field of SIP From Asterisk??
sip.conf useragent = myasteriskbox sdpsession = myasteriskbox On Wed, 2011-07-20 at 12:45 +0300, Israel Gottlieb wrote: user-agent could be set in sip.conf On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov abalas...@evaristesys.com wrote: On 07/20/2011 05:00 AM, Masood Ahmed wrote: Hello All, Is there any one who can help me to change the From field parameters in option packets, I have seen that in option packtes asterisk sends its own information,If you see the below option packet i have highlighted the asterisk word in from field and in from field tag how can i changed it Please let me know same as in User Agent. These are internally generated, so there is no way to modify them without a source-level change. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ilbc codec
Dear Sir, Can you confirm please if any version of asterisk does support ilbc 20ms instead of 30 ms sample frequency? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HELP - Client wants to reverse Asterisk Functionality
Hello, I'm putting Asterisk in to replace an existing IVR and that PBX system uses * to terminate number input instead of #. I thought it would be a matter of simply making a new app_read that replaced \# with \* but this didn't work. Any suggestions (besides bopping the client up side the head for wanting this?). Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call flow attached
Hello All, Could some one help me solve this. I want to configure asterisk as a MRF. The call flow is attached. The Asterisk server has to receive Invites send 200 OK with updated sdp. If Asterisk has to register with the proxy for this. I can do the necessary changes in my APP also. regards, Mohan attachment: call flow.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with System() application
Good afternoon,
I am trying to use the System() application but it is always
returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to run
this command:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt}
${exten});
This is the content of the /var/spool/asterisk/calllog/log.sh:
#!/bin/sh
#
#
TIME=$(date +%d-%m-%Y-%HH-%MM)
SOURCE=$1
DST=$2
echo $TIME - $SOURCE - $DST teste.log
I tried to insert some info direct into the file using echo but i've got
the same error.
Is there some secret to use this? haha
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[asterisk-users] 64 pickup groups
We have multiple customers running on a single Asterisk 1.4 installation and therefore require a large number of pickup groups. There seems to be a limitation of 64 call groups. Can anyone suggest how we work around this? For example is this limitation removed in a later version, is there a patch, are we approaching this in the wrong way? Any advice appreciated. attachment: winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with System() application
Are you able to execute: log.sh through the asterisk user?
On Wed, 20 Jul 2011 14:53:53 -0300, Antonio Modesto
mode...@isimples.com.br wrote:
Good afternoon,
I am trying to use the System() application but it is always
returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to run
this command:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt}
${exten});
This is the content of the /var/spool/asterisk/calllog/log.sh:
#!/bin/sh
#
#
TIME=$(date +%d-%m-%Y-%HH-%MM)
SOURCE=$1
DST=$2
echo $TIME - $SOURCE - $DST teste.log
I tried to insert some info direct into the file using echo but i've got
the same error.
Is there some secret to use this? haha
--
Jorge Gutiérrez
--
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[asterisk-users] ISAC and Asterisk
Are there any plans to include the ISAC codec in Asterisk? Is it possible or even desirable? Is ISAC open source (nothing indicates it is from the WebRTC website http://www.webrtc.org)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [SOLVED] Re: Problems with System() application
The problem was the directory which i was writing the logs, i put the
log file in /var/log/asterisk and it worked.
Thanks.
On Wed, 2011-07-20 at 13:03 -0500, Jorge Gutiérrez wrote:
Are you able to execute: log.sh through the asterisk user?
On Wed, 20 Jul 2011 14:53:53 -0300, Antonio Modesto
mode...@isimples.com.br wrote:
Good afternoon,
I am trying to use the System() application but it is always
returning APPERROR in the ${SYSTEMSTATUS} variable, I am trying to run
this command:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt}
${exten});
This is the content of the /var/spool/asterisk/calllog/log.sh:
#!/bin/sh
#
#
TIME=$(date +%d-%m-%Y-%HH-%MM)
SOURCE=$1
DST=$2
echo $TIME - $SOURCE - $DST teste.log
I tried to insert some info direct into the file using echo but i've got
the same error.
Is there some secret to use this? haha
--
Jorge Gutiérrez
--
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--
Atenciosamente,
Antônio Modesto
Gerente de TI
Praça Getúlio Vargas, 77 – Sala 308 – Centro
Santo Antônio do Monte – MG – CEP: 35560-000
Tel:(37) 3281-2800
Contato: isimp...@isimples.com.br
http://www.isimples.com.br
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Re: [asterisk-users] Problems with System() application
On Wed, 20 Jul 2011, Antonio Modesto wrote:
System(/bin/sh /var/spool/asterisk/calllog/log.sh ${FromExt} ${exten});
Specifying '/bin/sh' is not necessary.
The system() dialplan application calls the execl() system function with
the command '/bin/sh -c' so specifying '/bin/sh foo' results in sh running
sh running foo.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality
On Wed, 20 Jul 2011, Danny Nicholas wrote: I’m putting Asterisk in to replace an existing IVR and that PBX system uses * to terminate number input instead of #. How about an AGI executing some mix of get data, get option, stream file, or wait for digit and accumulate the digits yourself. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality
Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, July 20, 2011 4:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality On Wed, 20 Jul 2011, Danny Nicholas wrote: I’m putting Asterisk in to replace an existing IVR and that PBX system uses * to terminate number input instead of #. How about an AGI executing some mix of get data, get option, stream file, or wait for digit and accumulate the digits yourself. Or you can use WaitExten instead of Read. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, July 20, 2011 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality Sent from my Computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, July 20, 2011 4:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality On Wed, 20 Jul 2011, Danny Nicholas wrote: I?m putting Asterisk in to replace an existing IVR and that PBX system uses * to terminate number input instead of #. How about an AGI executing some mix of get data, get option, stream file, or wait for digit and accumulate the digits yourself. Or you can use WaitExten instead of Read. This sounded like a great idea, but I need numbers that are 4-16 digits in length, so I'm going to explore Steve's AGI suggestion (at least in 1.4, waitexten seems to be only good for 1 digit responses) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple SIP trunks between same pair of asterisk box
Hello, for billing purpose between a multitenant asterisk box and another asterisk, I am in the need to maintain multiple SIP trunks between them. Usually I use insecure=invite,port but I had to remove or the trunks will be selected based on IP address and not with username/password. I had to use the fromuser option or asterisk will try to authenticate the call using the CID and not the username, but this break the outbound CID of the client. Both are asterisk 1.6 Is there any other solution from multiple SIP trunks between two asterisk boxes? Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, July 20, 2011 3:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] HELP - Client wants to reverse Asterisk
Functionality
On Wed, 20 Jul 2011, Danny Nicholas wrote:
I?m putting Asterisk in to replace an existing IVR and that PBX system
uses * to terminate number input instead of #.
How about an AGI executing some mix of get data, get option, stream file, or
wait for digit and accumulate the digits yourself.
--
Great idea Steve (what else should I expect). I modified the agi-test.agi
that comes in the can to do just this, it just doesn't terminate when the
user presses * (still have to wait for the timeout)
Here it is if anyone cares -
#!/usr/bin/perl
use strict;
$|=1;
# Setup some variables
my %AGI; my $tests = 0; my $fail = 0; my $pass = 0;
my ($retvar, $rettime, $retlen) = @ARGV;
if (! $retvar) {
$retvar='digitacct';
$retlen=16;
$rettime=1;
}
if (! $retlen) {
$retlen=16;
$rettime=1;
}
if (! $rettime) {
$rettime=1;
}
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
print STDERR AGI Environment Dump:\n;
foreach my $i (sort keys %AGI) {
print STDERR -- $i = $AGI{$i}\n;
}
sub checkresult {
my ($res) = @_;
my $retval;
$tests++;
chomp $res;
if ($res =~ /^200/) {
$res =~ /result=(-?\d+)/;
if (!length($1)) {
print STDERR FAIL ($res)\n;
$fail++;
} else {
print STDERR PASS ($1)\n;
$pass++;
}
} else {
print STDERR FAIL (unexpected result '$res')\n;
$fail++;
}
}
print STDERR 1. Testing 'get data'...;
print GET DATA beep $rettime $retlen\\\n;
my $result = STDIN;
checkresult($result);
my $nresult=$result;
#my $rdigit=' ';
#while (length($nresult)16 $rdigit ne '*') {
# print WAIT DTMF 4000\n;
# $rdigit=STDIN;
# checkresult($rdigit);
# $nresult=$nresult.$rdigit if $nresult;
# $nresult=$rdigit if !$nresult;
# }
print STDOUT SET VARIABLE $retvar $nresult\n;
my $result2 = STDIN;
checkresult($result2);
print STDERR == Complete ==\n;
print STDERR $tests tests completed, $pass passed, $fail failed\n;
print STDERR ==
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Re: [asterisk-users] HELP - Client wants to reverse Asterisk Functionality
On Wed, 20 Jul 2011, Danny Nicholas wrote:
I?m putting Asterisk in to replace an existing IVR and that PBX system
uses * to terminate number input instead of #.
On Wed, 20 Jul 2011, Steve Edwards wrote:
How about an AGI executing some mix of get data, get option, stream
file, or wait for digit and accumulate the digits yourself.
On Wed, 20 Jul 2011, Danny Nicholas wrote:
I modified the agi-test.agi that comes in the can to do just this, it
just doesn't terminate when the user presses * (still have to wait for
the timeout)
On Wed, 20 Jul 2011, Steve Edwards wrote:
I was thinking more like (in PHP, sorry -- my Perl skills suck):
#!/usr/bin/php -q
?php
// initialize the AGI environment
require('phpagi.php');
$agi = new AGI();
// extract our parameters
$idx = 1;
$variable = $argv[$idx++];
$filename = $argv[$idx++];
$max_digits = $argv[$idx++];
$timeout = $argv[$idx++];
$terminator = $argv[$idx++];
// play the file and get the first digit
$dtmf = '';
$result = $agi-get_data($filename, $timeout, 1);
$key = $result['result'];
// loop for the remaining digits
while ((strlen($dtmf) $max_digits)
($key != $terminator))
{
$dtmf .= $key;
$result = $agi-wait_for_digit($timeout);
$key = chr($result['result']);
}
// set the channel variable
$agi-set_variable($variable, $dtmf);
?
And then, in your dialplan:
exten =
*,n,agi(read-with-terminator.php,RESPONSE,demo-congrats,10,1,*)
exten = *,n,verbose(1,${RESPONSE})
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] ISAC and Asterisk
On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote: Are there any plans to include the ISAC codec in Asterisk? Is it possible or even desirable? Is ISAC open source (nothing indicates it is from the WebRTC website http://www.webrtc.org)? What do you need it for? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] My Asterisk Box was hacked
Hi List, My asterisk box was hacked! Can anyone help on how do I secure my asterisk box, currently my box is installed with 2 NIC. 1st NIC is for LAN access and 2nd NIC has a public IP which is registered to our VoIP Provider. As I remember I already tried putting our Box on NAT but unfortunately due to some issue like call is dropped after 30 seconds and sometimes voice are not heard. Then we disable again the NAT. Your advise will be much appreciated. Thanks in advance. Regards, Malvin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
