Solved

Asterisk how to enable SRTP

Posted on 2010-08-30
24
2,965 Views
Last Modified: 2013-12-21
Hello ,
I want to know how to enable srtp in asterisk iam using Elastix 2.01 ??
Thanks...
0
Comment
Question by:tahasip
[X]
Welcome to Experts Exchange

Add your voice to the tech community where 5M+ people just like you are talking about what matters.

  • Help others & share knowledge
  • Earn cash & points
  • Learn & ask questions
  • 11
  • 7
  • 5
24 Comments
 
LVL 2

Expert Comment

by:xReaper
ID: 33557771
Hello; asterisk srtp is not in the box yet, so you dont have any way to enable it. There is some infos how to recompile asterisk with srtp, but you need to know asterisk really deeply.

Peoples installing elastix are generally now so much with tech details so i suggest the you try to do this some other way. i.e using vpn's.
0
 
LVL 32

Expert Comment

by:DrDamnit
ID: 33560131
You cannot enable it in Elastix. SRTP support must be compiled into the source code for it to work as described in my article:

http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Asterisk_/A_3233.html

You could (theorhetically) compile and replace the version of Asterisk that Elastix uses; however I highly doubt that would work. But... it's worth a try.
0
 

Author Comment

by:tahasip
ID: 33565118
Thanks both for your replay ,
Dear DrDamnit

I tried your post , every thing semmed to be ok , exept when try to do ((# Get asterisk
    svn co http://svn.digium.com/svn/asterisk/team/group/srtp/ asterisk))

I got url doesnot exist
So could you please tell me how to solve this problem??
0
Free learning courses: Active Directory Deep Dive

Get a firm grasp on your IT environment when you learn Active Directory best practices with Veeam! Watch all, or choose any amount, of this three-part webinar series to improve your skills. From the basics to virtualization and backup, we got you covered.

 
LVL 32

Expert Comment

by:DrDamnit
ID: 33567233
The SRTP branch has been merged to trunk for version 1.8, which is still in beta 4.

So, you can try asterisk 1.8 here:

http://svn.digium.com/svn/asterisk/trunk/

or you can download the SRTP branch that I just tarred up for you from my SRTP asterisk box here:

http://www.totalticketsystem.com/downloads/asterisk/srtp-asterisk.tar.gz
0
 
LVL 32

Expert Comment

by:DrDamnit
ID: 33569450
Do you have an update? Did you get your answer?
0
 

Author Comment

by:tahasip
ID: 33573954
i think iam confused now
do you mean that your post about srtp  
http://www.experts-exchange.com/Networking/Telecommunications/IP_Telephony/Asterisk_/A_3233.html 
must be for asterisk 1.8 that iam not using as i am using Elastix 2.0 with asterisk 1.6 ??

And for http://www.totalticketsystem.com/downloads/asterisk/srtp-asterisk.tar.gz this link what can i do with it , how can i use it ??
0
 

Author Comment

by:tahasip
ID: 33574595
hi,
I used this http://www.totalticketsystem.com/downloads/asterisk/srtp-asterisk.tar.gz and after that i have asterisk folder in /usr/src/ so, when try to compile
# Compile asterisk

cd /usr/src/asterisk

./configure

make

at the end i got error (((svn: This client is too old to work with working copy '.'; please get a newer Subversion client
   [CC] app_fax.c -> app_fax.o
app_fax.c: In function ¿transmit_audio¿:
app_fax.c:367: error: storage size of ¿fax¿ isn¿t known
app_fax.c:408: error: dereferencing pointer to incomplete type
app_fax.c:482: error: dereferencing pointer to incomplete type
app_fax.c:484: error: dereferencing pointer to incomplete type
app_fax.c:367: warning: unused variable ¿fax¿
app_fax.c: In function ¿transmit_t38¿:
app_fax.c:547: error: storage size of ¿t38¿ isn¿t known
app_fax.c:575: error: dereferencing pointer to incomplete type
app_fax.c:576: error: dereferencing pointer to incomplete type
app_fax.c:612: error: dereferencing pointer to incomplete type
app_fax.c:614: error: dereferencing pointer to incomplete type
app_fax.c:547: warning: unused variable ¿t38¿
make[1]: *** [app_fax.o] Error 1
make: *** [apps] Error 2)))

Also the same error when try to run
make install and make confige so, what to do ??
0
 
LVL 32

Expert Comment

by:DrDamnit
ID: 33577754
OK. That's not going to work because it relies on svn, whcih is not there. Also, that is Asterisk 1.4, and you need a minimum of 1.6, ergo, you're going to have to use 1.8 not the tarball I put up for you.

You'll have to use this link:

http://svn.digium.com/svn/asterisk/trunk/

you need to have subversion installed, and then the command is

cd /usr/src/
svn co http://svn.digium.com/svn/asterisk/trunk/ asterisk18
cd asterisk18
./configure
make menuselect
(configure menu select options, save, exit)
make
make install

Warning: this may get Elastix to work, it may break elastix entirely forcing a reinstall of the whole system.

...but it's your only hope of getting SRTP to work.
0
 

Author Comment

by:tahasip
ID: 33606246
Dear DrDamnit:

now when tried to make
make menuselect
(configure menu select options, save, exit)
make
make install

i got the this error
Shell options:
        -irsD or -c command or -O shopt_option          (invocation only)
        -abefhkmnptuvxBCHP or -o option
****
**** The configure script must be executed before running 'make'.
****               Please run "./configure".
****
make: *** [makeopts] Error 1

tell me what to do next to over come this error
0
 
LVL 2

Expert Comment

by:xReaper
ID: 33606375
As i said you will not able to do it.
You need to drop elastig and go from clean source install with asterisk 1.8, but its a suicide to put that in production. If you got some commmon sense drop it and search your solution in a different way.
0
 

Author Comment

by:tahasip
ID: 33606380
As i said you will not able to do it.
You need to drop elastig and go from clean source install with asterisk 1.8

How to do this ??
0
 
LVL 2

Expert Comment

by:xReaper
ID: 33606407
If you dont know hot to install clean asterisk from sources / svn  and configure all manually it will be difficult for you.

Do you know asterisk well ?
0
 

Author Comment

by:tahasip
ID: 33606421
no , i dont could you advise me ?
0
 
LVL 2

Expert Comment

by:xReaper
ID: 33606426
Well, before starting lets determinie if you has a reasont to use srtp.
Please describe your setup you want to achieve and why do you need srtp.

(network structure)
Thanks.
0
 

Author Comment

by:tahasip
ID: 33606449
ok ,
we need srtp as we face problem in one country here in middle east first we enable tls on astersik and the ATA is regestered ok, but now no voice is tansfered i can hear ring voice but no voice data is transported so, we think about srtp to sove this problem
0
 
LVL 2

Expert Comment

by:xReaper
ID: 33606626
I see, but how fast do you need a solution ? if you start reading yourself from 0 it will take you around 1-2 month to start doing things.
Have you considered to hire a professional ?
0
 

Author Comment

by:tahasip
ID: 33607600
no, if one or 2 monthes it is ok for us but please tell me how to start to be profitional in asterisk world
waiting your replay
0
 
LVL 2

Expert Comment

by:xReaper
ID: 33607697
Start reading this:
downloads.oreilly.com/books/9780596510480.pdf

and digging here:
http://www.voip-info.org/wiki/view/Asterisk+SRTP
0
 

Author Comment

by:tahasip
ID: 33607717
thanks for this but the pdf is for asteris 1.4 is ther any thing to 1.6 or this is enough ?
0
 
LVL 2

Expert Comment

by:xReaper
ID: 33607745
Its not a problem, you will learn the basis of asterisk in there.
0
 

Author Comment

by:tahasip
ID: 33609776
Thanks both xReaper: and DrDamnit:
now i have srtp working well
DrDamnit the missing part was ./configure --disable-xmldoc
after svn co http://svn.digium.com/svn/asterisk/trunk/ asterisk18
and every thing is fine and elastix also is fine thanks

You really profitional team

Thaaaaaaaaaaaanks
0
 

Author Comment

by:tahasip
ID: 33625523
Dear DrDamnit
Thanks for your replay , i want to tell you that first , iam using elastix version 2 with asterisk 1.6
i did all the steps at your post ,
svn co http://svn.digium.com/svn/asterisk/trunk/ asterisk18
./configure --disable-xmldoc
make
make configmenu
make install
make samples

now from elastix page i cannot see any extension all of them gone
and from FREE PBX i can see asterisk error
cant make reload or any thing

what i did i opened manager.conf file and amportal.conf and did some changes
then be able to make restore from FREE PBX and my system came back

by the way after finshed all the steps no sip trunk was registered

now is there any thing to do ,or i have to install asterisk 1.8(but it is still beta version) from beginning or i have to wait to elastix come with asterisk 1.8?or what
0
 
LVL 32

Accepted Solution

by:
DrDamnit earned 500 total points
ID: 33637029
I am not entirely familiar with the guts of elastix, but I can tell you the GUI systems like Elastix, Trixbox, SwitchVox, etc... use databases instead of flat files to control what is going on. They also have scripting that generates the GUI. So... when you re-installed Asterisk, it probably wiped out alot of your settings, which is why your extensions disappeared.

It appears that when you made the changes to ampportal.conf, it re-read the configs, and re-generated the files and settings for Elastix so that your extensions would come back.

My advice is to write down the procedure (and post here for posterity) so that you can re-produce this.

Once Asterisk 1.8 comes out with a non-beta or has other versions come out that are more stable, re-do this procedure.

Eventually, Elastix will come out with 1.8, but who knows when that will be....
0

Featured Post

Efficient way to get backups off site to Azure

This user guide provides instructions on how to deploy and configure both a StoneFly Scale Out NAS Enterprise Cloud Drive virtual machine and Veeam Cloud Connect in the Microsoft Azure Cloud.

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

Suggested Solutions

Title # Comments Views Activity
VOIP Setup through a Watchguard BOVPN 4 104
ASA 5505 Slowing Internet 11 160
Digium IP D40 Phones - Cisco Voice VLAN-ing 7 102
send SMS from desktop FREE 14 91
Hey there Heard about jingle, the add on for XMPP that enables point to point audio between two XMPP clients. No server config necessary. Actually quite a cool feature. However, how good is it if you can not use those voice capabilities to do a P…
If your business is like most, chances are you still need to maintain a fax infrastructure for your staff. It’s hard to believe that a communication technology that was thriving in the mid-80s could still be an essential part of your team’s modern I…
Nobody understands Phishing better than an anti-spam company. That’s why we are providing Phishing Awareness Training to our customers. According to a report by Verizon, only 3% of targeted users report malicious emails to management. With compan…
Finds all prime numbers in a range requested and places them in a public primes() array. I've demostrated a template size of 30 (2 * 3 * 5) but larger templates can be built such 210  (2 * 3 * 5 * 7) or 2310  (2 * 3 * 5 * 7 * 11). The larger templa…

735 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question