Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk
Thank you very much Tim, this looks quite promising! Just sad, that once again one has to compile it instead of provided packages ): But it's probably worth the work :-) 2012/8/29 Tim Nelson tnel...@rockbochs.com - Original Message - Yeah, I noted that too, but besides that it seems like it is exactly what I am looking for. I am especially confused that there's no hint like hey, buy our new product, just EOL. So let's say I am looking for an alternative to this. And unfortunately I have to add it's for private use and I therefore need a free solution, which probably restricts the selection ): Well, anything better than checking logs by hand would be already a good start :-) Sorry for digging up a zombie thread (Jun 20th or thereabouts)... I just stumbled upon Homer SIP Capture. It's 100% open source, and looks to be what you're in search of. Have a look: http://www.sipcapture.org/ --Tim -- _ -- 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
Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Regards. On Tue, Aug 28, 2012 at 10:47 PM, Bryant Zimmerman brya...@zktech.comwrote: I would install both dahdi and libpri. I brought up a 12.2 RC-2 VM on hyper-v Windows 8 and followed our standard asterisk build and have no issues yet but we have not run full testing to confirm. Also a point of not 12.2 is RC for the next 8 days or so. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 -- *From*: Gopalakrishnan N gopalakrishnan...@gmail.com *Sent*: Tuesday, August 28, 2012 1:13 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 If I don't need to install dahdi hardware, is it really I need to have libpri installed? Regards. On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote: Check Jason Parker’s post from today and see if you skipped any of the preliminary build steps. It is possible that something like libpri is biting you. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Tuesday, August 28, 2012 11:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 I tried that too, what happens is asterisk is loading but after that if I try to start any one module for example chan_sip.so, asterisk hangs. Regards. On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote: Extensions/trunks. Another thought is that you might make your modules.conf not load anything to start with so you can eliminate a rogue module as the problem. Just change autoload=yes to autoload=no. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 11:47 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi danny, Are you talking about modules or sip extensions and dahdi extensions because its a fresh installation also it doesn't have dahdi interface, I am just trying to have only ip side. Regards On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote: I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and 10 SP2). My advice would be to try to start the box with as few SIP/DAHDI channels as possible to begin with and add as you get things stable. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N *Sent:* Monday, August 27, 2012 8:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2 Hi Patrick, With other OS it works like charm. Only with OpenSuse, I am facing this issue, since I have a situation to stick with OpenSuse, I am struggling in this. Regards. On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 27-08-12 08:25, Gopalakrishnan N wrote: This is really tuff working with OpenSuse. I am clueless how to sort our this. Maybe switch to a different distribution? I have used CentOS and RHEL for years without any problems and as far as I know both debian and ubuntu should work without problems too. Regards, Patrick -- _ -- 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 -- _ -- Bandwidth and Colocation
Re: [asterisk-users] Click-to-call software in a hosted environment
On Wednesday 29 August 2012, Carlos Alvarez wrote:
For any of you doing hosted PBX service on Asterisk, do you have a reliable
and secure click to dial solution? Particularly for Outlook, but since
about 20% of our customers use Mac OS, I'd love to hear about some that
work on that too.
This is my generic works-anywhere click-to-call script. It should work in
conjunction with any software that allows you to specify an external command
to call a number (we have tried it with Kontact and it works beautifully).
You just have to issue a wget command to fire a CGI script on the server.
Place this in your Asterisk server's /usr/lib/cgi-bin folder, call it
make_call.pl and chmod 755 make_call.pl:
8
#!/usr/bin/perl -w
use strict;
use DBI;
my ($web, $input_buffer, $name, $value, %parameters);
my ($ip, $ext, $tel);
my $dbh = DBI-connect(DBI:mysql:database=phonestuff;host=localhost, root,
);
my $sth_get_ext = $dbh-prepare(SELECT ext FROM extensions WHERE pc_ip LIKE
?);
foreach (split//, $ENV{'QUERY_STRING'}) { # GET items
tr/+/ /;
($name,$value) = split /=/, $_;
$name =~ s/%(..)/pack'c', hex $1/eg;
$value =~ s/%(..)/pack'c', hex $1/eg;
$parameters{$name} = $value;
};
read STDIN, $input_buffer, $ENV{CONTENT_LENGTH}; # POST items
foreach (split//, $input_buffer) {
tr/+/ /;
($name,$value) = split /=/, $_;
$name =~ s/%(..)/pack 'c', hex $1/eg;
$value =~ s/%(..)/pack 'c', hex $1/eg;
$parameters{$name} = $value;
};
print Content-type: text/plain\n\n;
$tel = $parameters{tel} || ;
unless ($ext = $parameters{ext}) {
$sth_get_ext-execute($ENV{REMOTE_ADDR});
if ($sth_get_ext-rows) {
($ext) = $sth_get_ext-fetchrow_array;
};
$sth_get_ext-finish;
};
if ($ext) {
print Calling from '$ext' to '$tel'.\n;
open CALLFILE, /tmp/asterisk_$$.call;
print CALLFILE --STOP--;
Channel: SIP/$ext
Context: outgoing
extension: $tel
Priority: 1
CallerId: $ext
--STOP--
close CALLFILE;
system mv /tmp/asterisk_$$.call
/var/spool/asterisk/outgoing/${ext}_${tel}.call;
}
else {
print Go away, we don't know who you are. (try ext=something)\n;
};
$dbh-disconnect;
exit;
8
You also need a database `phonestuff` with a table `extensions` relating PC IP
addresses (in `pc_ip`) to extension numbers (in `ext`).
Now if your software works anything like Kontact, it will want you to specify
a command to place a call and can substitute placeholders in this command.
So, give this as the command:
wget -o /dev/null http://ip.of.asterisk.server/cgi-bin/make_call.pl?tel=%N
(Test it in an xterm, omitting the -o /dev/null, with something like your
mobile number or another extension.)
Licence: This program is copyright (C) 2012 by A J Stiles. You are permitted
and even encouraged to distribute this program, modified or unmodified, in
Source Code form whether or not accompanied by a binary executable version
provided that this notice accompanies every copy. Binary distribution without
Source Code constitutes a violation of copyright.
--
AJS
Answers come *after* questions.
--
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[asterisk-users] change channel variable to a user chosen value during a call
Hi, I'm in the following usecase: A customer calls in to an asterisk box and the call is answered by an employee. During this call, the employee has to set the CDR(accountcode) for the channel from his phone. Is this possible with asterisk? What I've tried so far is to add a dynamic feature in the [applicationmap] section of features.conf which executes a Macro (I know it's not recommended according to the comment in the sample config file). Executing this macro works, but when using the Read application asterisk does not hear the DTMF digits I enter. Anyone else ran into the same situation and has any recommendations? Regards, Frederic -- _ -- 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] Asterisk hangs while starting in OpenSuse 12.2
On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk That should give you some low level info what's going on. More info about strace and available options can be found in: $ man strace Regards, Patrick -- _ -- 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] Spa3102 info about tones an frecuency for Brasil's analog line
Hello all, Does anybody know the correct setings for tones and frecuency for a Brazil's analog line to be configured on a Linksys 3102? Busy tone, disconect tone, ,etc. Thanks in advance, Ismael. -- _ -- 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] Install AsteriskNow
Thanks for your message, If I understand correctly. I have to copy the Asterisk installation code to AsteriskNow installation. And setup complier in AsteriskNow and do the make sample? On Aug 30, 2012, at 12:37 AM, Richard Mudgett rmudg...@digium.com wrote: I am trying to install an AsteriskNow. When system boot up, there are two options, To install with Asterisk 1.8 and FreePBX type 1 ENTER To install with Asterisk 1.8 only type 2 ENTER If I want to install Asterisk 1.8 only for example. After asterisk is install, I found the /etc/asterisk/ is empty. I am wondering if any good way that I could have some sample configurations. Run make samples That will copy all of the sample config files from ./configs into /etc/asterisk with appropriate removal of the .sample from the filenames. Richard -- _ -- 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
Re: [asterisk-users] change channel variable to a user chosen value during a call
More information please - Asterisk version and are you using realtime? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frederic Van Espen Sent: Thursday, August 30, 2012 4:35 AM To: asterisk-users Subject: [asterisk-users] change channel variable to a user chosen value during a call Hi, I'm in the following usecase: A customer calls in to an asterisk box and the call is answered by an employee. During this call, the employee has to set the CDR(accountcode) for the channel from his phone. Is this possible with asterisk? What I've tried so far is to add a dynamic feature in the [applicationmap] section of features.conf which executes a Macro (I know it's not recommended according to the comment in the sample config file). Executing this macro works, but when using the Read application asterisk does not hear the DTMF digits I enter. Anyone else ran into the same situation and has any recommendations? Regards, Frederic -- _ -- 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
Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote: On 08/30/2012 09:45 AM, Gopalakrishnan N wrote: Hi, I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I am not using any virtualbox, still i struck in loading the modules. Please do not top post. Install strace and then start asterisk with the command: # strace asterisk Asterisk will fork into the background and the process you trace will exit. strace -f asterisk #? strace asterisk -f #? Just in case you wonder, 'asterisk -f strace' will not work as you might have expected from the above examples. Nither will '-f strace asterisk'. '-U asterisk ' may also come in handy. -- 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
Re: [asterisk-users] change channel variable to a user chosen value during a call
On Thu, 2012-08-30 at 08:23 -0500, Danny Nicholas wrote: More information please - Asterisk version and are you using realtime? Currently running asterisk 1.8.13.0 and not using realtime. -- _ -- 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] AST-2012-012: Asterisk Manager User Unauthorized Shell Access
Asterisk Project Security Advisory - AST-2012-012 Product Asterisk Summary Asterisk Manager User Unauthorized Shell Access Nature of Advisory Permission Escalation Susceptibility Remote Authenticated Sessions SeverityMinor Exploits Known No Reported On July 13, 2012 Reported By Zubair Ashraf of IBM X-Force Research Posted OnAugust 30, 2012 Last Updated On August 30, 2012 Advisory ContactMatt Jordan mjordan AT digium DOT com CVE NameCVE-2012-2186 Description The AMI Originate action can allow a remote user to specify information that can be used to execute shell commands on the system hosting Asterisk. This can result in an unwanted escalation of permissions, as the Originate action, which requires the originate class authorization, can be used to perform actions that would typically require the system class authorization. Previous attempts to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought to do so by inspecting the names of applications and functions passed in with the Originate action and, if those applications/functions matched a predefined set of values, rejecting the command if the user lacked the system class authorization. As reported by IBM X-Force Research, the ExternalIVR application is not listed in the predefined set of values. The solution for this particular vulnerability is to include the ExternalIVR application in the set of defined applications/functions that require system class authorization. Unfortunately, the approach of inspecting fields in the Originate action against known applications/functions has a significant flaw. The predefined set of values can be bypassed by creative use of the Originate action or by certain dialplan configurations, which is beyond the ability of Asterisk to analyze at run-time. Attempting to work around these scenarios would result in severely restricting the applications or functions and prevent their usage for legitimate means. As such, any additional security vulnerabilities, where an application/function that would normally require the system class authorization can be executed by users with the originate class authorization, will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has been updated to reflect that the AMI Originate action can result in commands requiring the system class authorization to be executed. Proper system configuration can limit the impact of such scenarios. The next release of each version of Asterisk will contain, in addition to the fix for the ExternalIVR application, an updated README-SERIOUSLY.bestpractices.txt file. Resolution Asterisk now checks for the ExternalIVR application when processing the Originate action. Additionally, the README-SERIOUSLY.bestpractices.txt file has been updated. It is highly recommended that, if AMI is utilized with accounts that have the originate class authorization, Asterisk is run under a defined user that does not have root permissions. Accounts with the originate class authorization should be
[asterisk-users] AST-2012-013: ACL rules ignored when placing outbound calls by certain IAX2 users
Asterisk Project Security Advisory - AST-2012-013 ProductAsterisk SummaryACL rules ignored when placing outbound calls by certain IAX2 users Nature of Advisory Unauthorized use of system SusceptibilityRemote Authenticated Sessions Severity Moderate Exploits KnownNone Reported On 07/27/2012 Reported By Alan Frisch Posted On 08/30/2012 Last Updated OnAugust 30, 2012 Advisory Contact Matt Jordan mjordan AT digium DOT com CVE Name CVE-2012-4737 Description When an IAX2 call is made using the credentials of a peer defined in a dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are not applied to the call attempt. This allows for a remote attacker who is aware of a peer's credentials to bypass the ACL rules set for that peer. Resolution The ACL rules for peers defined in an ARA backend are now honored. Users of chan_iax2 should upgrade to the corrected versions; apply a provided patch; or define their IAX2 peers outside of an ARA backend in a static configuration file. Affected Versions ProductRelease Series Asterisk Open Source 1.8.x All versions Asterisk Open Source 10.x All versions Certified Asterisk 1.8.11 All versions Asterisk Digiumphones 10.x.x-digiumphones All versions Asterisk Business EditionC.3.x All versions Corrected In Product Release Asterisk Open Source 1.8.15.1, 10.7.1 Certified Asterisk 1.8.11-cert7 Asterisk Digiumphones 10.7.1-digiumphones Asterisk Business Edition C.3.7.6 Patches SVN URL Revision http://downloads.asterisk.org/pub/security/AST-2012-013.1.8.diff Asterisk 1.8 http://downloads.asterisk.org/pub/security/AST-2012-013.10.diff Asterisk 10 Links https://issues.asterisk.org/jira/browse/ASTERISK-20186 Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/AST-2012-013.pdf and http://downloads.digium.com/pub/security/AST-2012-013.html Revision History Date Editor Revisions Made 08/27/2012 Matt Jordan Initial Revision Asterisk Project Security Advisory - AST-2012-013 Copyright (c) 2012 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form. -- _ -- 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] change channel variable to a user chosen value during a call
Is realtime an option for you to install? If it is you could develop a web interface that allows you to put the customer account number in, or even integrate it into your existing customer management system. Depending on the scale of what you're doing though, this might be overkill. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frederic Van Espen Sent: Thursday, 30 August 2012 11:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] change channel variable to a user chosen value during a call On Thu, 2012-08-30 at 08:23 -0500, Danny Nicholas wrote: More information please - Asterisk version and are you using realtime? Currently running asterisk 1.8.13.0 and not using realtime. -- _ -- 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] failed to extend from 512 to 676 message on console
Hello, Asterisk Version 1.6.2.9 on below hardware. We are using 100 Realtime SIP extensions. CPU : 1 x Intel® Core-i5 3.3 GHz. RAM : 4 GB DDR-3 SDRAM Hard Disk : 500 GB Hard Disk For last few days, getting below messages on asterisk cli. We googled to find the solution for this but could not locate the preventive steps. failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 failed to extend from 512 to 676 Thanks,Kamlesh -- _ -- 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
