Re: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance cannot encode gsm via MixMonitor

[Patrick Laimbock]

Hi Stefan,

> Hi all
> 
> I maintain the above - it was set up by an external party with whom relations 
> have now been severed by my employer.
> 
> Quite early after the deployment it became evident that all .gsm audio files 
> produced on this virtual instance at Azure via MixMonitor are corrupt.
[snip]

Is the CentOS 7 installation/image the same across your bare-metal hosts and 
the one on azure? AFAIK there is still no official CentOS 7 image provided by 
the CentOS Project on the azure marketplace. Instead it's created by a third 
party [1]. So there may be differences that could cause issues. On your azure 
host, check the repo files in /etc/yum.repos.d/. If the mirrorlist/basurl 
points to openlogic or roguewave than it's a third-party image. IIRC Amazon and 
GCP have official CentOS 7 images provided by the CentOS Project. Maybe try one 
of those to see if the issue persists? Alternatively create your own CentOS 7 
VM from the official CentOS 7 repositories using kickstart and try that on 
azure.

Best, Patrick

[1] 
https://azuremarketplace.microsoft.com/en-us/marketplace/apps/RogueWave.CentOS76?tab=Overview

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Re: [asterisk-users] Detecting DoS attacks via SIP

[Patrick Laimbock]

Hi Mike,

On 15-08-17 21:37, mdiehl wrote:

Hi all,

Lately, I've seen an increase in the number of attacks against my system from the 
so-called "Friendly Scanner."  When one of these script kiddies targets my 
server, all I see for symptoms is a few of my trunks become lagged due to server load and 
a stream of messages on the console that resemble this:

[snip]

I have to turn on sip debugging to find out who's hitting me.  However, I can't 
just leave it on because it would kill my logging system.

So, how are other people handling this?  Is there an AMI event I want watch 
for?  I watch for PeerStatus, but since there's no actual peer in the attack, I 
don't seem to get an event from AMI.

Any ideas?


You can block sipvicious/friendly scanner in iptables with something like:

-A INPUT -p udp --dport 5060 -m string --string "friendly-scanner" 
--algo bm -j DROP


You can also look at xtables with geoip to drop countries (per 
destination port) that should not connect to your Asterisk box. It's a 
big hammer but it works really well.


Or put a proxy like Kamailio or OpenSIPS in front of the Asterisk box. 
That's what the telco's/service providers do.


HTH,
Patrick

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Re: [asterisk-users] fail2ban Asterisk 13.13.1

[Patrick Laimbock]

On 02-03-17 13:52, Bryant Zimmerman wrote:

John V

Are you using pjsip? We are have several test servers and  I just
checked my /etc/fail2ban/filter.d/asterisk.conf and it is not updated
for pjsip implementations.  Looking at the security log files and the
regex I noticed that some items are being banned but others are not due
to changes in the messages for pjsip.
Anyone got an updated asterisk.conf for fail2ban.


The latest upstream version of asterisk.conf can be found here:

https://github.com/fail2ban/fail2ban/blob/0.10/config/filter.d/asterisk.conf

This commit mentions improved pjsip support:

https://github.com/fail2ban/fail2ban/commit/f85fb45b29768f687546ba25f805977cf00b6e43

HTH,
Patrick



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Re: [asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?

[Patrick Laimbock]

On 03-01-17 19:06, Joshua Colp wrote:

On Fri, Dec 30, 2016, at 05:04 AM, Kevin Long wrote:



Hello,

I am using asterisk 14.2 and PJSIP,  with TLS transport.

I’m sure I’m doing something wrong here ..


In 2 distinct softphone clients (Bria and Groundwire),  I am able to
register successfully,  and place a SIP call, with no certificate
warnings. But shortly after I place that first call and hang up,  I
receive a certificate name mismatch error in the softphone,  the error
presenting me with the *IP adddress* of my Asterisk server,  not the
hostname, and of course the TLS certificates only have the hostname, not
the IP, and I have configured the soft phone to use the hostname, not the
IP, to connect.


I’m guessing there is some currently unset hostname setting within
asterisk/pjsip that is defaulting to sending the IP in the sip messages,
and then when the soft phone tries to make a new tls sip connection to
asterisk,  perhaps to signal to asterisk that the call is complete,  it
then connects to the IP instead of the hostname, and the mismatch occurs
?


This might be the Contact header. Right now there is no ability to
configure this to a hostname instead of an automatically determined IP
address.


Thank you for your feedback Joshua. Does "right now" mean that this will 
be fixed in the (near) future? Should I file a Jira ticket?


Thanks,
Patrick

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Re: [asterisk-users] Asterisk 13.5 and higher (asterisk 13.7.2) quitting

[Patrick Laimbock]

Hi Travis,

On 04-03-16 15:23, Ryan, Travis wrote:

I start asterisk 13.7.2 and it dies before I can rasterisk into it. I’ve
tried getting a coredump, but it doesn’t coredump.  I know there are a
lot of errors in the log below, but most of those just say it’ll not
load a module, and no big deal.

When launching from commandline (not service script) here is what happens.

http://pastebin.com/3GFe6fG9


Two things:

[Mar  3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module: Error 
loading module 'res_monitor.so': 
/usr/lib/asterisk/modules/res_monitor.so: undefined symbol: __ast_beep_stop
[Mar  3 15:19:37] WARNING[8439]: loader.c:553 load_dynamic_module: Error 
loading module 'res_ari_events.so': 
/usr/lib/asterisk/modules/res_ari_events.so: undefined symbol: 
stasis_app_register_all


Undefined symbol errors are not good. Not sure why that's just a 
WARNING. Maybe something went wrong during the build? The output of the 
build should show you more information. In the mean time try disabling 
these two modules just to see if that clears up the problem.


[Mar  3 15:19:39]   == Parsing '/etc/asterisk/extensions.conf': Found
[Mar  3 15:19:39] WARNING[8439]: config.c:2228 config_text_file_load: 
Unterminated comment detected beginning on line 386


That needs fixing.

HTH,
Patrick


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Re: [asterisk-users] cdr_odbc: Error in ExecDirect: -1

[Patrick Laimbock]

On 01/13/16 14:48, Vitor Mazuco wrote:

Hi everybody!

I'm trying to install a CDR Viewer https://github.com/g613/asterisk-cdr-viewer

I'ts work well, but my problem is to make Asterisk store to MySQL using ODBC

When I make a call the CLI returns for me

See the log:
== Using SIP RTP CoS mark 5
 -- Executing [2021@ramais:1] Dial("SIP/2020-",
"SIP/2021,60,tT") in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/2021
 -- SIP/2021-0001 is ringing
> 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
[Jan 13 11:31:07] NOTICE[2279][C-]: res_rtp_asterisk.c:4441
ast_rtp_read: Unknown RTP codec 95 received from '(null)'
 -- SIP/2021-0001 answered SIP/2020-
> 0x7fd3b4004eb0 -- Probation passed - setting RTP source
address to 192.168.25.100:8000
> 0x7fd3f8014240 -- Probation passed - setting RTP source
address to 192.168.25.49:35528
> cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] WARNING[2279][C-]: res_odbc.c:604
ast_odbc_direct_execute: SQL Execute error! Verifying connection to
asterisk [asterisk-connector]...
> cdr_odbc: Error in ExecDirect: -1
[Jan 13 11:31:08] ERROR[2279][C-]: cdr_odbc.c:177 odbc_log:
CDR direct execute failed


See my res_odbc.conf

[asterisk]
enabled = yes
dsn = asterisk-connector
username = root
password = 100567
pooling = no
limit = 1
pre-connect = yes

What can be happened?

Thank in advanced.


Just a guess but try setting "pooling" to yes and "limit" to a higher value.

Best,
Patrick

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Re: [asterisk-users] Signaling ringing on other extension

[Patrick Laimbock]

On 12/30/15 12:24, Luca Bertoncello wrote:

Ishfaq Malik  schrieb:


Do you have a link to the user guide for your exact phone model?


Unfortunately not...
I have a Thomson ST2022, but I can just find in Internet manual for the
ST2030...


The administrator manual can be found at:
http://www.manualslib.com/manual/909341/Thomson-St2020.html?page=5

To download click the green Download button at the top.

In the right column there is also a link to the User Guide.

Cheers,
Patrick

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Re: [asterisk-users] Fritzbox 7490

[Patrick Laimbock]

On 08-06-15 19:00, Christian wrote:



Hi,
Sorry if off topic, but is anyone here on this list using it?
I am currently searching for a good router for my home network wich supports 
SIP.
Many thanks!


I use a 7360 and it works ok but if the 7490's firmware is anything like 
the 7360 then be prepared for some fixing before it works with Asterisk 
 SIP on port 5060:


http://blog.laimbock.com/2014/03/27/how-to-make-asterisk-work-behind-fritz-box/

HTH,
Patrick

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Re: [asterisk-users] Asterisk y Ldap

[Patrick Laimbock]

On 16-12-14 14:00, Dario Estupinan wrote:

Como integrar asterisk con Ldap.??


https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver

Best,
Patrick

ps this mailing list uses the English language

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Re: [asterisk-users] ICE consuming High CPU

[Patrick Laimbock]

On 05-12-14 08:25, Mayank Kumar Gour wrote:

Any help will be appreciated.


Help us help you by providing much much much more information then you 
have right now. http://www.catb.org/esr/faqs/smart-questions.html


HTH,
Patrick


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Re: [asterisk-users] Asterisk 13 LDAP

[Patrick Laimbock]

On 04-12-14 11:23, Phil Daws wrote:

Is there still an LDAP driver as do not see it in the CentOS 6 repository ?


AFAICT in EPEL only Asterisk 1.8.32.1 has one:
http://koji.fedoraproject.org/koji/buildinfo?buildID=594932

If you need a newer version then you'll have to build it yourself or 
find a repo you trust that carries Asterisk 13 with the LDAP module.


The OpenLDAP developers recommend to always use the latest OpenLDAP 
release and skip the distro shipped one. If you are going to build 
Asterisk packages yourself then you may want to build them against 
OpenLDAP 2.4.40 and use that. Or switch to CentOS7 which has an almost 
current OpenLDAP. Both Symas and the LTB Project have current OpenLDAP 
RPMs for EL6 ( EL7):


https://symas.com/products/symas-openldap-directory/
http://ltb-project.org/wiki/

HTH,
Patrick

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Re: [asterisk-users] Asterisk 11.9.0 crash and restart

[Patrick Laimbock]

On 21-10-14 08:54, 為近 吉摩（情報システム本部）- Tamechika Yoshikiyo - 
wrote:

Hi,

My Asetrisk restarted after to output following warning message.

  [Oct 16 15:59:58] WARNING[17102][C-8e34]: chan_sip.c:4696
update_provisional_keepalive: Unable to cancel schedule ID 738278.  This
is probably a bug (chan_sip.c: update_provisional_keepalive, line 4696).


That looks similar to this bug:
https://issues.asterisk.org/jira/browse/ASTERISK-21387

Why not install the latest Asterisk version (11.13.1) so you have all 
the latest fixes and can see if the bug is still present?


Also your Asterisk 11.9.0 version is subject to the POODLE vulnerability 
for which a fix is available in 11.13.1.


http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.13.1.tar.gz

HTH,
Patrick

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Re: [asterisk-users] bristuff-0.4.0-RC4-xr7

[Patrick Laimbock]

On 21-10-14 12:36, Ray Image wrote:
[snip]

Thanks for your response. As far as I can see I have no choice but use
Zaptel as that is downloaded as part of the bristuff script. If there is
a better way of getting a HFC PCI card to play nicely with Asterisk
please let me know :-)


The HFC based Asterisk B410P and Hx8 + B400M cards are supported by 
DAHDI. If you use something with a HFC-S chip then you can use the 
sources below which add support for HFC-S to DAHDI. Note that they may 
not use the latest DAHDI version so you may have to figure out a way to 
get the patches applied to current DAHDI. The one time I tried the 
dahdi-hfcs stuff it seemed to work fine (very light usage only).


http://sourceforge.net/projects/dahdi-hfcs/
http://www.openvox.cn/pub/drivers/dahdi-linux-complete/

HTH,
Patrick

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Re: [asterisk-users] Asterisk Crashes Randomly with Cepstral Swift TTS

[Patrick Laimbock]

Hi Bryan,

On 10/18/2014 11:47 PM, Bryan Burroughs wrote:
 All,
 
 Has anyone seen this before? This appears to be a Swift or app_swift
 bug. I'm having a difficult time finding any information or support on this.

I haven't used app_Swift with Cepstral but iirc it wasn't deemed very
stable.

 Asterisk version:
 Asterisk 11.6-cert4 built by asterisk @ ivrd02 on a x86_64 running Linux
 on 2014-08-11 13:55:25 UTC
 OS:
 Linux livrp03 2.6.32-431.11.2.el6.x86_64 #1 SMP Mon Mar 3 13:32:45 EST
 2014 x86_64 x86_64 x86_64 GNU/Linux

If you are not tied to the certified Asterisk version then perhaps try
using the latest Asterisk version (currently 11.13.0).

 When Asterisk crashes, the backtrace always looks something like the
 following:

[snip]

 The out of bounds line looks like it may be pointing to the issue.
 
 *argv = {0x0, 0xb9b0 Address 0xb9b0 out of
 bounds, 0x0}*

Have you tried contacting the app_swift developer and/or filed a bug at
https://issues.asterisk.org/jira/secure/Dashboard.jspa ?

 Should I look into using another TTS engine?

You could try UniMRCP which sits between Asterisk and Cepstral replacing
app_swift: http://unimrcp.org

HTH,
Patrick

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Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13

[Patrick Laimbock]

Hi Rainer,

On 15-09-14 09:07, Rainer Piper wrote:

Hi,

Info !!! not a question !!!

the pjsip logger is different:

[Sep 15 07:33:27] NOTICE[65267] res_pjsip/pjsip_distributor.c: Request
from '1001 sip:1001@81.20.137.222' failed for '85.25.197.23:5071'
(callid: 1bfa1fcfee1e20dbe9bbbcac5d7bdffc) - No matching endpoint found

and here the RegEx for fail2ban to catch this log:

|NOTICE.* .*: Request from '.*' failed for 'HOST(:[0-9]{1,5})?' (.*) -
No matching endpoint found


Thanks for sharing. If you use github it would be nice if you could 
submit a pull request so that it becomes part of the Asterisk rules in 
the next Fail2ban version (0.9.1).


https://github.com/fail2ban/fail2ban/pulls

HTH,
Patrick

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Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13

[Patrick Laimbock]

On 15-09-14 17:22, Rainer Piper wrote:

Hi Patrick,

github done ;-)


Thanks!


what is HTH ???


Hope this/that helps

http://www.internetslang.com/
http://www.urbandictionary.com/define.php?term=internet%20slang

HTH :)
Patrick


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Re: [asterisk-users] Asterisk secure fine tune - stop attack

[Patrick Laimbock]

On 04-09-14 16:44, motty cruz wrote:

Hi All,
I see this kind of attack on our Asterisk Server, do you know how to
block that IP?

[Sep  4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite:
Call from '' (213.136.81.166:9306 http://213.136.81.166:9306) to
extension '34422' rejected because extension not found in context 'default'.


Have a look at Fail2ban:
http://www.fail2ban.org/wiki/index.php/Main_Page

HTH,
Patrick

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Re: [asterisk-users] Setup Own IP PBX Server

[Patrick Laimbock]

On 01-09-14 12:31, Chandran Manikandan wrote:
[snip]

I have installed Freepbx server and tried to configure sip extension.
It's working fine.


A better place for FreePBX related questions and to get help is:
http://community.freepbx.org/
Or hire their professional FreePBX support:
http://www.freepbx.org/support-and-professional-services

If you want to learn more about Asterisk in general then a good start is 
to first read Asterisk: The Definitive Guide, 4th Edition and go 
through the wiki at http://wiki.asterisk.org.


HTH,
Patrick

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Patrick Laimbock]

On 02-09-14 20:18, Khalid Touati wrote:

so it seems Asterisk Versions does not support video I guess


On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the 
Bria app on Android and iPhone. With SELinux and the firewall 
temporarily disabled I couldn't get it to work with either H264 or VP8.


HTH,
Patrick

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Patrick Laimbock]

On 02-09-14 21:15, Eric Wieling wrote:

As long as you are NOT transcoding video should work in Asterisk.


Both apps were configured with identical (codec) settings so I don't see 
how it would require transcoding. If you did get it to work I would 
appreciate it if you could tell me which clients you used, the Asterisk 
version, the OS and the relevant Asterisk config.


Thanks,
Patrick



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 20:18, Khalid Touati wrote:

so it seems Asterisk Versions does not support video I guess


On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the
Bria app on Android and iPhone. With SELinux and the firewall
temporarily disabled I couldn't get it to work with either H264 or VP8.

HTH,
Patrick




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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Patrick Laimbock]

On 02-09-14 22:52, Eric Wieling wrote:

A co-worker was doing video, I dislike video.  The phones were Polycom VVX, The 
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only 
two video codecs enabled.


Thanks Eric. The obvious difference is that your co-worker was using 
H.263/H.263p while Bria uses H.264 or VP8. videosupport=yes was present 
in my sip.conf so it might be the codec. Time for more tinkering.


Thanks,
Patrick

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Re: [asterisk-users] RDNIS with tel: vs. sip: header

[Patrick Laimbock]

On 28-08-14 11:57, Positively Optimistic wrote:

Has anyone had success patching chan_sip.c so that Asterisk will
recognize the tel: header for RDNIS information?


  exten = get_in_brackets(tmp);
 if (!strncasecmp(exten, sip:, 4)) {
 exten += 4;
 } else if (!strncasecmp(exten, sips:, 5)) {
 exten += 5;
 } else {
 ast_log(LOG_WARNING, Huh?  Not an RDNIS SIP header
(%s)?\n, exten);
 return -1;
 }

Audiocodes Mediant 2000 devices send this header as a tel:...

*[Aug 28 02:25:42] WARNING[1283][C-1574] chan_sip.c: Huh?  Not an
RDNIS SIP header (tel:41068558XX)?*
*
*
*(number obscured for privacy purposes)*


Not a dev but have you tried something like this (hope the formatting 
stays sane):


exten = get_in_brackets(tmp);
  if (!strncasecmp(exten, sip:, 4)) {
exten += 4;
  } else if (!strncasecmp(exten, tel:, 4)) {
exten += 4;
  } else if (!strncasecmp(exten, sips:, 5)) {
exten += 5;
  } else {
ast_log(LOG_WARNING, Huh?  Not an RDNIS SIP header (%s)?\n, exten);
return -1;
  }

HTH,
Patrick

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Re: [asterisk-users] Understanding local channels

[Patrick Laimbock]

On 25-08-14 17:06, Mitch Claborn wrote:

Can someone point me to a good tutorial / explanation of local
channels?  I've been using them without really understanding what is
going on, and we all know how dangerous that is!

I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but
I'm just not quite getting it.


How about the info on the Asterisk wiki:

https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels

On the left side there's a menu with examples and modifiers.

HTH,
Patrick

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Re: [asterisk-users] Dahdi CAPI migration

[Patrick Laimbock]

On 11-08-14 11:09, Toney Mareo wrote:


  Hello

The answers to your questions are:

1, OS
CentOS release 5.5 (Final)


That version is ancient and full of security holes. It is recommended to 
at least update to CentOS 5.10 + updates. That's assuming there are 
Trixbox kmod-dahdi-linux* RPMs for CentOS 5.10.



Trixbox installed at: Autogenerated by /usr/sbin/dahdi_genconf on Fri Nov 25 
18:03:26 2011


Trixbox CE no longer exists and is no longer supported. Why continue 
using it?



2, Kernel
Linux 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:53:10 EST 2010 i686 i686 i386 
GNU/Linux

3, Packages

asterisk16-dahdi.i3861.6.0.26-1_trixboxinstalled
dahdi-firmware.noarch2.0.0-1_centos5   installed
dahdi-firmware-oct6114-064.noarch1.05.01-1_centos5 installed
dahdi-firmware-oct6114-128.noarch1.05.01-1_centos5 installed
dahdi-firmware-tc400m.noarch MR6.12-1_centos5  installed
dahdi-firmware-vpmadt032.noarch  1.07-1_centos5installed
dahdi-linux.i386 2.3.0.1-1_trixbox installed
dahdi-tools.i386 2.3.0-1_trixbox   installed
dahdi-tools-doc.i386 2.2.0-4_trixbox   installed
kmod-dahdi-linux.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5
kmod-dahdi-linux-xen.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5
dahdi-linux-devel.i386   2.3.0.1-1_trixbox trixbox28
kmod-dahdi-linux-PAE.i6862.3.0.1-1_trixbox.2.6.18_164.11.1.el5
libpri.i386  1.4.10.2-1_centos5installed
libpri-devel.i3861.4.10.2-1_centos5trixbox28
asterisk16.i386  1.6.0.26-1_trixboxinstalled
kmod-mISDN.i686  1.1.7.2-4_centos5.2.6.18_164.11.1.el5
kmod-mISDN-xen.i686  1.1.7.2-3_centos5.2.6.18_164.11.1.el5
mISDN.i386   1.1.7.2-4_centos5 installed
mISDNuser.i386   1.1.7.2-2_centos5 installed
asterisk-chan_misdn.i386 1.4.22-3  trixbox
kmod-mISDN-PAE.i686  1.1.7.2-3_centos5.2.6.18_164.11.1.el5
mISDN.i686   1.1.7-27  trixbox
mISDN-debuginfo.i686 1.1.7-24  
trixboxaddons
mISDN-devel.i686 1.1.7-27  trixbox
mISDN-devel.i386 1.1.7.2-4_centos5 trixbox28
mISDN-kmod-base.i686 1.1.7.2-1_centos5.2.6.18_128.1.10.el5
mISDN-modules.i686   1.1.7-27.2.6.18_92.1.18.el5   trixbox
mISDNuser-debuginfo.i386 1.1.7-15  
trixboxaddons
mISDNuser-devel.i386 1.1.7.2-2_centos5 trixbox28


All ancient, with many (security) bugs and no longer supported.


Asterisk 1.6.0.26-FONCORE-r78, Copyright (C) 1999 - 2010 Digium, Inc. and 
others.

4, What do you mean with the OS-es were clones ...? Did you create an
image of the old Trixbox machine and installed that on the new machine?

It means that they are Xen virtual machines, exact bit by bit vm clones so they 
should have all the same configuration files, run the exact same Xen kernels. 
What complicates things a bit, and probably the cause of my errors is Xen's PCI 
passthrough. The only reason why I use something so obsolete like Xen is just 
this feature otherwise I would be using kvm, vmware, virtualbox or whatever 
virt technologies but for those you must have vt(d) hardware support and the 
machine I dealing with here doesn't have this, neither the old one.


Right.


5, Lsdadhi (this is on the first, working machine)

### Span  1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
   1 BRIClear   (In use) (SWEC: MG2)
   2 BRIClear   (In use) (SWEC: MG2)
   3 BRIHardware-assisted HDLC  (In use)
### Span  2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS
   4 BRIClear   (In use) (SWEC: MG2)
   5 BRIClear   (In use) (SWEC: MG2)
   6 BRIHardware-assisted HDLC  (In use)
### Span  3: B4/0/3 B4XXP (PCI) Card 0 Span 3 AMI/CCS RED
   7 BRIClear   (In use) (SWEC: MG2)  RED
   8 BRIClear   (In use) (SWEC: MG2)  RED
   9 BRIHardware-assisted HDLC  (In use)  RED
### Span  4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS
  10 BRIClear   (In use) (SWEC: MG2)
  11 BRIClear   (In use) (SWEC: MG2)
  12 BRIHardware-assisted HDLC  (In use)


Ok.


6, Asterisk logs (new machine when it failed)

full.4:[Aug  7 12:39:58] WARNING[1654] chan_dahdi.c: Unable to specify channel 
1: No such device or address
full.4:[Aug  7 12:39:58] ERROR[1654] chan_dahdi.c: Unable to open channel 1: 

Re: [asterisk-users] Dahdi CAPI migration

[Patrick Laimbock]

On 08-08-14 10:09, Toney Mareo wrote:

Hello

Thank you for your response. I thought it could be easier moving the old card 
to the new machine and using the DAHDI driver. Unfortunately my first attempt 
for this failed. The card shows up in the original machine as:


dahdi_hardware -v

pci::00:00.0 wcb4xxp+ 1397:08b4 Junghanns QuadBRI ISDN card


IIRC the wcb4xx module is correct for this card.


00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller 
[HFC-4S] (rev 01)
 Subsystem: Cologne Chip Designs GmbH HFC-4S [IOB4ST]
 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- 
Stepping- SERR- FastB2B-
 Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- 
MAbort- SERR- PERR-
 Interrupt: pin A routed to IRQ 20
 Region 0: I/O ports at 9400 [size=8]
 Region 1: Memory at f2841000 (32-bit, non-prefetchable) [size=4K]
 Capabilities: [40] Power Management version 2
 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA 
PME(D0-,D1-,D2-,D3hot-,D3cold-)
 Status: D0 PME-Enable- DSel=0 DScale=0 PME-

The second machine recognized it as well and as I said the OS-es were clones 
(so all drivers, settings should be indentical), still after start the DAHDI 
driver loaded but didn't work from Asterisk:


What do you mean with the OS-es were clones ...? Did you create an 
image of the old Trixbox machine and installed that on the new machine?


Which versions are you using of:
the OS?
dahdi-linux?
dahdi-tools?
libpri?
asterisk?

Are you using mISDN too?


Running dahdi_cfg:  DAHDI_SPANCONFIG failed on span 1: No such device or 
address (6)
No hardware timing source found in /proc/dahdi, loading dahdi_dummy


In general it could be that the card is not recognized or that there's a 
conflict with another kernel module or your dahdi config is incorrect or 
the udev rules are missing or wrong, etc.



Any ideas how to proceed from this point?


Since you have provided not very much information one can only guess. 
Only by providing as much information as possible can you help us help you.


So please answer all the questions, provide the versions as asked, give 
the output of lsdahdi, check the logfiles (/var/log/messages etc) for 
dahdi info, check the dahdi config and blacklist files and provide that 
with anything else you can think of.


AFAIK the Junghanns QuadBRI ISDN card should work fine with a recent 
DAHDI so why do you want to change to an AVM C4 card that requires 
chan_capi that hasn't seen any recent development and does not support 
Asterisk 11 or later? Asterisk 1.8 will only be supported for another 
year and then you are stuck again with an obsolete system that no longer 
gets any (security!) updates.


If you can, why not stay with the Junghanns card and put a fresh CentOS 
6.5 + updates on the new machine together with the latest version of 
dahdi-linux, dahdi-tools, libpri and asterisk. And then copy your old 
dialplan over to the new machine and make the required changes so it 
works with the latest Asterisk 11. Or you could install the latest 
FreePBX iso on the new machine if you prefer a GUI.


HTH,
Patrick

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Re: [asterisk-users] Dahdi CAPI migration

[Patrick Laimbock]

Hi Toney,

Comments inline.

On 07-08-14 12:10, Toney Mareo wrote:

Hello Folks,
I looking to migrate a pbx from one server to another. The original server has 
this ISDN card:

00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller 
[HFC-4S] (rev 01)

The new server:
00:00.0 I2O: Digital Equipment Corporation StrongARM DC21285 (rev 04)  AVM 
ISDN Controller C4


Now I'm not fully aware of both's cards functions since the manuals are very brief. What 
I heard is that the later card does not support some NT commands what I might going to 
need at this migration. I can't really say if one card is better than the 
other one.
Unfortunately the two uses completely different drivers. The first card uses 
dahdi, the second uses capi. I want this migration go as smooth as possible 
with the least downtime so I looking for some help
maybe someone has more experience with these cards.


In the past I have used Eicon Diva Server cards with chan_capi for years 
and they worked great. I



I also read something about CAPI uses completely different dial plans.
So all the Asterisk configurations are migrated from the old server to the new 
one.


That is correct although the differences/changes you need to make are 
limited. Adjusting the Dial command will go a long way.



By following this guide:
http://www.asteriskguru.com/tutorials/avm_c4.html

I have the C4 modul loaded. My asterisk box is Asterisk 1.6.0.26-FONCORE-r78 
(Trixbox).


That Asterisk version is rather old and assuming that's the old Trixbox 
CE it is without security updates since 2012. I recommend you use a 
fresh install of something like CentOS 6.5 + all updates and Asterisk 
11.11.0 (or later if available) with the latest dahdi-linux, dahdi-tools 
and libpri releases. Also get the latest chan_capi from here: 
ftp://ftp.chan-capi.org/chan-capi/ The version with -HEAD in the name 
has the latest fixes and is the one I always used.



How can I see that this C4 card is really working from asterisk?


Once you have the AVM C4 kernel module loaded and Asterisk with 
chan_capi is installed and capi is enabled in the Asterisk config, start 
Asterisk and then you should have a 'capi' command in the CLI. Executing 
it should show you info about the status of your ISDN channels.



What is the difference between the chan_dahdi.conf and chan_capi.conf?


One is for DAHDI supported cards and the other is for CAPI cards like 
the Eicon Diva Server and the AVM C4.



Can't I just tell it somewhere to use the new card and I don't have to touch 
the existing dialplans etc?


Nope.

HTH,
Patrick

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Re: [asterisk-users] Security Architecture or Security Evaluations Docs?

[Patrick Laimbock]

On 28-07-14 12:28, Jeffrey Walton wrote:
[snip]


Is there anything that includes the development process? I'm
interested in the secure development items and testing.


Info about the development of Asterisk can be found here:
http://asterisk.org/community/developers
https://wiki.asterisk.org/wiki/display/AST/Development

Development related questions can best be asked on the asterisk-dev 
mailing list or on irc.freenode.net in #asterisk-dev.


HTH,
Patrick




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Re: [asterisk-users] Security Architecture or Security Evaluations Docs?

[Patrick Laimbock]

On 26-07-14 14:23, Jeffrey Walton wrote:

Does anyone know of Security Architecture or Security Evaluations
documents that I could read?

Searching is turning up no hits. For example,
http://www.google.com/#q=security+evaluation+site:asterisk.org and
http://www.google.com/#q=security+architecture+site:asterisk.org.


Assuming security+evaluation refers to Common Criteria, I'm not aware 
of any Common Criteria initiatives in relation to Asterisk (nor 
FreeSWITCH, OpenSIPS, Kamailio, Yate or any other Open Source VoIP 
project I'm aware of). Asterisk is a toolbox with many flexible building 
blocks and not a product like Cisco CallManager with pre-defined 
features set in stone. As such it doesn't really make sense to get 
Asterisk certified, if possible at all. It would be like trying to 
certify C or Python. If EALx certification is your requirement then have 
a look at the CallManager as iirc it's EAL1 certified.


Re asterisk+architecture, Asterisk Security related best practices are 
described here:

http://svn.asterisk.org/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt

HTH,
Patrick

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Re: [asterisk-users] VoIP over 3G/4G Data

[Patrick Laimbock]

On 18-07-14 17:59, Tech Dude wrote:

What are the recommended settings to successfully implement VoIP over
3G/4G data connection. Assume we are talking about using Polycom phones,
and the 3G/4G data connection comes from a Cradlepoint router that is
plugged in with AC power and has high gain antennas. The device will be
stationary, so we will not have to worry about tower handoff’s breaking
the connection. This will be for fixed wireless.

I have read to use G.729 codec, and TCP for signaling to bypass
firewalls. Besides that, what other settings are recommended?  Changes
in MTU? Changes in voice payload ms? Is there a better codec to use?
Header compression?


Use TLS/SRTP so the carrier can't do packet inspection/snooping and mess 
with or block your VoIP connections. They might throttle/block port 5060 
and 5061 anyway in which case you should use different ports.


I use Asterisk 11 with TLS/SRTP, G.722 and Android phones (4G, 
CSipSimple or Bria), iPhones (4G, Bria) and Polycom phones. G.722 on 
inter-office calls has been working great so no need for G.729 and 
Asterisk uses standard alaw/ulaw for regular PSTN calls. I use TCP, have 
made no MTU changes and use standard 20ms voice payload.


Packet loss, latency and jitter are the enemy. Your router might be top 
notch but if the cell tower is overloaded and experiences too much 
packet loss, delays of more than 150ms and lots of jitter than you may 
get crappy sound quality. If possible, get some prepaid 4G sim cards for 
your router from different carriers and test which carrier consistently 
provides the best signal, least delay, packet loss and jitter.


HTH,
Patrick

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Re: [asterisk-users] Asterisk and LDAP

[Patrick Laimbock]

On 18-06-14 23:06, Linus Lüssing wrote:

Hi,

I'm trying to get Asterisk running with LDAP to be able to
authenticate sip user registrations. I'm using Asterisk
(1.8.13.1~dfsg1-3+deb7u3) on a Debian server.

Unfortunately I wasn't successful so far.

My res_ldap.conf looks like this (so pretty minimal):
---
[_general]
;url=ldaps://ldap.chaotikum.org
url=ldap://ldap.chaotikum.org
protocol=3
basedn=dc=chaotikum,dc=org

[sip]
name = uid


IIRC the recommendation in the latest Asterisk book is to use only a-z, 
numerics (0-9) and underscore. So if you have [t...@chaotikum.org] in 
sip.conf then that might not work because of the '@'.


You can easily test this by adding a peer [test_1234] (so with the 
recommended syntax) and add it to your LDAP server with a password and 
then check if it registers.


HTH,
Patrick

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Re: [asterisk-users] Asterisk and LDAP

[Patrick Laimbock]

On 20-06-14 15:05, Linus Lüssing wrote:
[snip]

having [test_phone_120d] in my sip.conf works fine. Ah wait - do
I need to have a user both in LDAP and sip.conf and the only
thing LDAP can do for me is the authentication/password checking?


As far as I know, yes :)

Cheers,
Patrick

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Re: [asterisk-users] SSL/TLS weakness impact on Asterisk authentication

[Patrick Laimbock]

On 10-06-14 23:44, Michelle Dupuis wrote:

After reading about the  2 major SSL (and TLS?) weaknesses discovered
this year, I was wondering how it affects asterisk.

Does the SIP authentication use TLS - or something that was recently
broken?  Is there a risk of exposing passwords?


Asterisk' SIP authentication uses a digest. See 
http://tools.ietf.org/html/rfc3261 for more info (20.6 and onwards).


That does not mean that the recent OpenSSL issues have no impact on 
Asterisk. They do if you configure SIP to use TLS transport or enable 
TLS for other parts (for example AMI). So it's highly recommended to 
install the updated OpenSSL packages containing the fixes.


My Asterisk packages link dynamically against the OpenSSL libraries. 
Assuming your packages do the same then, once you have updated the 
OpenSSL packages to the latest ones with the fixes and restart Asterisk, 
you should be good to go.


While the recent OpenSSL issues don't directly expose your account 
passwords, the Heartbleed bug can expose (parts of) the private key used 
by TLS. Once the Men in Black have your private key its possible to 
setup a Man (in Black) in the Middle attack and sniff those passwords. 
See http://heartbleed.com/


Unless you want to mess around with the Men in Black and leave your 
system vulnerable to attack, you should install all security updates 
ASAP and then restart the services that rely upon them.


HtH,
Patrick

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Re: [asterisk-users] SIP Softphone

[Patrick Laimbock]

On 09-06-14 08:52, Giles Coochey wrote:

On 08/06/2014 22:01, Mark Robinson wrote:

Hello,

can someone recommend a good and free Softphone for Windows which does
not display advertisments inside the program?



Has anyone tried MicroSIP?
http://www.microsip.org/


Nope but if it doesn't meet your needs then maybe have a look at Jitsi 
https://jitsi.org/ or Linphone https://www.linphone.org/


I prefer the client to have at least the following features:

Security:
- TLS
- SRTP
- ZRTP

Codecs:
- G722
- G729

Fight NAT (if IPv6 is not an option):
- STUN
- TURN
- ICE

Cheers,
Patrick

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[asterisk-users] iPhone TLS reg problem: FILE * open failed

[Patrick Laimbock]

Hi,

I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to register 
with TLS to an Asterisk 11.10.0 box. The registration fails and I see 
this in the Asterisk console:


== Problem setting up ssl connection: error:: 
lib(0):func(0):reason(0)
[Jun  8 15:33:39] WARNING[8555]: tcptls.c:274 handle_tcptls_connection: 
FILE * open failed!


Anyone know what that error means? The source code does not tell me 
much. FWIW the same setup works fine with an Android phone.


Thanks!
Patrick

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Re: [asterisk-users] iPhone TLS reg problem: FILE * open failed

[Patrick Laimbock]

On 06/08/14 16:32, Mehroz Ashraf wrote:

Random guess:

1. Make sure you have enabled tls/ssl for that particular user/extension

2. Check the iphone date/time, should be within the certificate validation.


Thank you for you suggestions. The iPhone has the correct date  time 
which is also within the date/time range of the client certificate. The 
extension is TLS enabled. Using the same details on an Android phone 
work fine.


I just tried another iPhone client called Join and, in spite of a 
terrible interface, TLS registration works and TLS/SRTP calls work too. 
Guess it's a problem in Linphone.


Cheers,
Patrick


On Jun 8, 2014 6:50 PM, Patrick Laimbock patr...@laimbock.com
mailto:patr...@laimbock.com wrote:

Hi,

I'm trying to setup an iPhone 4S (iOS 7.1.1) with Linphone to
register with TLS to an Asterisk 11.10.0 box. The registration fails
and I see this in the Asterisk console:

== Problem setting up ssl connection: error::
lib(0):func(0):reason(0)
[Jun  8 15:33:39] WARNING[8555]: tcptls.c:274
handle_tcptls_connection: FILE * open failed!

Anyone know what that error means? The source code does not tell me
much. FWIW the same setup works fine with an Android phone.

Thanks!
Patrick

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Re: [asterisk-users] Get last dialed number in a context?

[Patrick Laimbock]

On 03-06-14 11:31, Stefan Gofferje wrote:

Hi,

I would like to implement an auto-redial function in a context. The idea
is about like this:

Dial a number
Hear busy
Hangup
Pick up again
Dial a code like *123
= jumps into a context which redials until callresult is not busy

Maybe like this:

[autoredial]
exten = s,1,Set(number=${CHANNEL(lastdialed)})
exten = s,2,Dial(SIP/${number}@account,60,g)
exten = s,3,Wait(15)
exten = s,4,GotoIf( [ ${DIALSTATUS} = BUSY ]?2)
exten = s,5,Hangup

For that I'd need to somewhere get the last dialed number from the
channel/line I'm initiating the call from. Is something like this
already implemented?


Have you looked at Call Completion Supplementary Services (CCSS)?
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243096

Cheers,
Patrick

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Re: [asterisk-users] Kernel and DAHDI

[Patrick Laimbock]

On 12-05-14 09:30, Thorsten Göllner wrote:

That's correct. When you update the kernel package youhave also to
recompile dahdi package.


AFAIK that's not true for RHEL6/CentOS6 as the EL6 kernels are ABI 
compatible so you don't need to recompile DAHDI when there's a new EL6 
kernel. When updating a RHEL5/CentOS5 kernel I always had to recompile 
DAHDI for the new EL5 kernel.


HtH,
Patrick

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Re: [asterisk-users] One mailbox for multiple extensions with individual greetings

[Patrick Laimbock]

On 11-05-14 05:10, John T. Bittner wrote:

Why don't you use the voicemail copy feature?
Create 3 mailboxes 1234, 6789 and 2000 for the shared.

VoiceMail(1234@default2000@default,su)
VoiceMail(6789@default2000@default,su)

Set both 1234 and 6789 to email the voicemail to a fake email address and 
delete after email.
A copy of the message for each will be dropped into 2000 and deleted from the 
original box.


Thanks John. I'll give that a try.

Cheers,
Patrick


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[asterisk-users] One mailbox for multiple extensions with individual greetings

[Patrick Laimbock]

Hi,

Is there a way in Asterisk 11 to use a single voicemailbox for multiple 
extensions while still hearing each extension's individual greeting?


Use case: someone has 2 numbers and wants all voicemail messages for 
both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER 
you would hear the person at extension 1234 is unavailable and the 
message would be stored in mailbox mymailbox and when dialing 6789 and 
NOANSWER you would hear the person at extension 6789 is unavailable 
and that message would also be stored in mailbox mymailbox. The user 
then dials an extension to reach mymailbox and hears all messages for 
both the 1234 and 6789 numbers.


I think I can solve it by symlinking 
/var/spool/asterisk/voicemail/default/6789/INBOX to 
/var/spool/asterisk/voicemail/default/1234/INBOX (and the other 
directories too) but it would be nice if this could be done within the 
dialplan.


If that's not possible, would adding an extension option to 
app_voicemail.c solve this by decoupling the extension from the mailbox?


Thanks for any pointers.

Cheers,
Patrick

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Re: [asterisk-users] AMR installation error

[Patrick Laimbock]

On 30-04-14 12:50, [Digital^Dude] ® wrote:

make gives this:


IIRC Digium's policy is that there's no support on this list for 
patented technologies like AMR which are possibly not officially 
licensed. Obviously to prevent any legal liability.


HTH,
Patrick

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Re: [asterisk-users] Asterisk support for h.324m

[Patrick Laimbock]

On 29-04-14 20:41, [Digital^Dude] ® wrote:

Hello,

If anyone has successfully compiled asterisk with:
app_rtsp
codec_amr
mp4_play
mp4_save
app_transcode
h324m_call

Please share the versions of OS software, and libraries used.
Lets make this thread useful so that all tried and tested video
resources of asterisk can be found in one place for ease of access and
later reference.


I haven't but the guys you could talk to are the (former Fontventa?) 
folks at http://www.medooze.com/products/h324m-stack.aspx


Source code/binaries can be viewed/downloaded at:

http://sourceforge.net/p/asteriskvideo/
http://sourceforge.net/projects/mcumediaserver/

The old Fontventa AMR patch does not apply to Asterisk 11. I tried to 
port it to Asterisk 11 but couldn't get a call going. If you are a 
developer focused on Asterisk 11 and want to have a look at the AMR 
patch let me know and I'll email it to you.


HTH,
Patrick

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Re: [asterisk-users] Trunk issue

[Patrick Laimbock]

On 28-04-14 19:49, Haley,Scott A wrote:

Now I am getting Permission denied.


Have you checked if SELinux is blocking the app? Any blockage should 
show up as an 'AVC' in /var/log/audit/audit.log You can temporarily set 
SELinux to permissive with 'setenforce 0' and check if the problem goes 
away.


HTH,
Patrick

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Re: [asterisk-users] Trunk issue

[Patrick Laimbock]

On 28-04-14 20:13, Haley,Scott A wrote:

That seemed to fix it. Thanks to everyone.


https://bugzilla.redhat.com/show_bug.cgi?id=1092150

HTH,
Patrick

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Re: [asterisk-users] Asterisk and SRTP

[Patrick Laimbock]

On 04/05/2014 07:56 PM, William Wu wrote:

Hi experts,

I am trying Asterisk SRTP in my environment, and find that when
Asterisk is behind a NAT, the audi/video UDP ports opened for SRTP relay
by Asterisk are local ports on the Asterisk server, media from the two
clients out of the NAT (for example from Internet) can not reach the
ports, and thus the two client can not establish the secure call via
Asterisk. I have set up a STUN server and configured in rtp.conf, but
seems Asterisk does not do STUN before it opens ports for SRTP. BTW,
Non-SRTP call can work though.

   Anyone can give advice on how to make SRTP work in such an env?


I have no problems with a TLS/SRTP call between a GSM with CSipSimple 
and Asterisk 11.8.1 behind NAT. Have you configured the NAT options in 
sip.conf?


externip=...
localnet=...
nat=...

You may also need to add/change the options below. Check the sip.conf 
example file to see what these options do and use what's best for your 
situation.


canreinvite=no
directmedia=no
directrtpsetup=no

HTH,
Patrick

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[asterisk-users] CLI command to see if SRTP is active?

[Patrick Laimbock]

Hi,

I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI 
command to see if SRTP is active on a channel/call. I went through sip 
show ... and core show channel... and did not see any mentioning of SRTP 
while there is an SRTP call active.


Thanks,
Patrick

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[asterisk-users] Problem with TLS/SRTP with Asterisk 11.8.1

[Patrick Laimbock]

Hi,

I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 
on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local 
wifi. The phone seems to register but directly after that things fall 
apart (turning SELinux off made no difference):


*CLI -- Registered SIP 'encrypted' at 10.0.0.137:58079
Saved useragent CSipSimple_crespo-19/r2330 for peer encrypted
SSL certificate ok
  == Problem setting up ssl connection: error:14094410:SSL 
routines:SSL3_READ_BYTES:sslv3 alert handshake failure
[Mar 24 21:20:42] WARNING[28466]: tcptls.c:272 handle_tcptls_connection: 
FILE * open failed!
[Mar 24 21:20:45] NOTICE[28460]: chan_sip.c:29584 sip_poke_noanswer: 
Peer 'encrypted' is now UNREACHABLE!  Last qualify: 0

SSL certificate ok
  == Problem setting up ssl connection: error:14094410:SSL 
routines:SSL3_READ_BYTES:sslv3 alert handshake failure
[Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection: 
FILE * open failed!

-- Unregistered SIP 'encrypted'

sip.conf looks like this:

[general]
context=guest
allowguest=no
allowoverlap=no
allowtransfer=no

bindaddr=0.0.0.0:5060
udpbindaddr=0.0.0.0:5060
tcpenable=no

tlsenable=yes
tlsbindaddr=0.0.0.0

tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt

tlscipher=ALL
tlsclientmethod=tlsv1

transport=udp

preferred_codec_only=no
disallow=all
allow=ulaw
language=en
trustrpid=no
dtmfmode=rfc2833
videosupport=no
alwaysauthreject=yes
directmedia=no
jbenable = yes
jbforce = no

[encrypted]
type=friend
secret=1234
context=internal
callerid=Encrypted 1002
host=dynamic
qualify=yes
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
transport=tls
encryption=yes


$ ls -l /etc/asterisk/keys
total 28
-rw-r--r--. 1 asterisk asterisk 1204 mrt 24 16:16 asterisk.crt
-r--r-. 1 asterisk asterisk  887 mrt 24 16:16 asterisk.key
-r--r-. 1 asterisk asterisk 2091 mrt 24 16:16 asterisk.pem
-rw-r--r--. 1 asterisk asterisk 1736 mrt 24 16:16 ca.crt
-r. 1 asterisk asterisk 3311 mrt 24 16:16 ca.key
-rw-r--r--. 1 asterisk asterisk 1208 mrt 24 16:20 nexus.crt

The certs were created with ast_tls_cert as described in the tutorial. I 
created a nexus.p12 for the phone and imported it before configuring 
CSipSimple.


Does anyone know what's wrong? Pointers much appreciated.

Thanks,
Patrick

[0] https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial

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Re: [asterisk-users] Problem with TLS/SRTP with Asterisk 11.8.1

[Patrick Laimbock]

On 24-03-14 21:28, Patrick Laimbock wrote:
[snip]

   == Problem setting up ssl connection: error:14094410:SSL
routines:SSL3_READ_BYTES:sslv3 alert handshake failure
[Mar 24 21:20:56] WARNING[28467]: tcptls.c:272 handle_tcptls_connection:


So others may find the fix: make sure the server and client certificates 
have the proper keyUsage. The ast_gen_tls script does not set them and 
this caused the handshake/verification to fail.


The client certificate needs something like:
keyUsage = digitalSignature, keyEncipherment
extendedKeyUsage = clientAuth

The server certificate needs something like:
keyUsage = digitalSignature, keyEncipherment
extendedKeyUsage = serverAuth

HTH,
Patrick

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Re: [asterisk-users] dahdi + dlink du128ta

[Patrick Laimbock]

On 03/14/2014 07:53 AM, binary dreamer wrote:

hello everyone,
I do have a usb ISDN modem that I would like to make it work with dahdi.
is it possible?


No.


I am running debian 7, with dahdi 2.9, asterisk 11.8
dahdi cannot find it at the moment, unless there is something else to be
done.


Try building  installing the stuff from http://misdn.eu  Works fine 
with a USB TA with HFC chipset last time I tested it.


HTH,
Patrick

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Re: [asterisk-users] dahdi + dlink du128ta

[Patrick Laimbock]

On 03/15/2014 10:15 PM, binary wrote:

i have tried the misdn from git. my problem is that it needs LCR and it
fails to get installed


Then you need to fix that. AFAIK there is no other way to use a USB ISDN 
TA than via mISDN/LCR.


HTH,
Patrick

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Re: [asterisk-users] Linux call router

[Patrick Laimbock]

On 11-03-14 12:15, binary dreamer wrote:

hello there,
I am facing an issue with misd/misdnuser/lcr in the system
I am running debian 7 and I managed to install from git misdn/misdnuser
but in lcr I am getting:
chan_lcr.c: In function 'load_module': chan_lcr.c:3520:24: warning:
assignment makes pointer from integer without a cast [enabled by
default] make[2]: *** [chan_lcr.po] Error 1 make[2]: Leaving directory
/usr/src/lcr' make[1]: *** [all-recursive] Error 1 make[1]: Leaving
directory /usr/src/lcr' make: *** [all] Error 2 root@voyage:/usr/src/lcr#
could someone help me please?


I have not built lcr in a while (on CentOS) but iirc: make sure you get 
from http://misdn.eu/ the latest isdn4k-utils, mISDN, mISDNuser and lcr. 
Then build in that order. For lcr I had these build requirements: 
autoconf automake libtool libtiff-devel mISDNuser-devel ncurses-devel 
openssl-devel


If you still can't figure it out perhaps ask on the ISDN4Linux mailing 
list: https://www.isdn4linux.de/mailman/listinfo/isdn4linux


Cheers,
Patrick

ps don't build as root, it's bad practice.

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Re: [asterisk-users] Asterisk crashes at meetme kick all

[Patrick Laimbock]

On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote:

Dear Forum,
I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk
crashed while executing “meetme kick all” CLI command from manager
interface. The link says the issue has been closed however I am not able
to identify in which release of asterisk this issue has been fixed.
Please help.
_https://issues.asterisk.org/jira/browse/ASTERISK-15741_


AFAICT this issue has not been fixed due to inactivity. Note the 
Suspended due to lack of activity remark. Also the 1.6 version 
mentioned in the bugreport is EOL. Version 10.0.0 you mentioned is also 
EOL so any bugreport you file against version 10.0.0 will not be acted 
upon unless you can reproduce it with the latest Asterisk version 11.x.x 
or 12.x.x.


I recommend you upgrade to an Asterisk LTS (long term support) version 
like the latest 11.x.x (currently 11.7.0). For more information about 
Asterisk LTS versions go to:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

If you still see a bug when running Asterisk 11.x.x (or 12.x.x) you can 
report it at the Asterisk issue tracker at:


https://issues.asterisk.org/jira/secure/Dashboard.jspa

Before filing a bug please read the information at:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines#AsteriskIssueGuidelines-Howtoreportabug

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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

[Patrick Laimbock]

On 25-01-14 06:26, Amit wrote:

Thanks for response.
How do I derive the requirement? I need to size IO system to record multiple 
calls concurrently.


I'm not aware of 400+ calls being recorded succesfully on an Asterisk 
box. If there is it probably has tons of RAM, enterprise grade SSDs or 
15K RPM FC/SAS drives in a battery backed RAID setup or a fast SAN 
saving the calls in native format (via a tmpfs) with the transcoding 
probably done on another box.



I ran test with following configuration
Quad Core Xeon with 4GB RAM


Add more RAM and much much more if you are going to use tmpfs.


250GB SATA disk (No RAID)


Well you get the performance you pay for. CentOS comes with various 
utilities that allow you to analyze that.



Linux (CentOS 5.9)


Imo CentOS 6.5 (x86_64) has better performance.


Asterisk 1.8.20


In 9 months Asterisk 1.8 will only get security fixes. I would use 
Asterisk 11. It will get regular bug fixes for a much longer time.



I failed to record more than 80 calls.


Hardly surprising.


If I run test with simple IVR, I achieved 400+ calls with same server.


A simple IVR is not the same as call recording. The comparison makes as 
much sense as saying that copying to /dev/null is faster than to a disk.



So write seem to be an issue.
Is there any way to tune / optimize / configure for better write performance?

I am not sure if I need to post this query on developers list? Please guide...


No, this is a user question and does not belong on the developer list.

Since you seem to work for a call center business perhaps investigate a 
commercial solution like Orecx (I have no affiliation):


http://www.orecx.com/OrecX-for-Asterisk.php

HTH,
Patrick

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