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  • Asteriks +TLS+RSTP Encrypt on centos

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    this guide was gathered from official asteriks wikies (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM(CentOS6/RedHatEnterpriseLinux6, https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics, https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial, https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics) and tested in VPS server with Centos 6 OS

    first of all install libsrtp and libsrtp-devel

    yum install libsrtp libsrtp-devel

    If the packages cannot be found make sure you have epel repository installed.

    Adding the repository

    rpm -Uvh http://packages.asterisk.org/centos/6/current/i386/RPMS/asterisknow-version-3.0.0-1_centos6.noarch.rpm
    yum update


    yum install asterisk asterisk-configs --enablerepo=asterisk-11

    install dahdi

    yum install dahdi-linux dahdi-tools libpri

    again upgrade

    yum update

    Basic configure add user accounts into the /etc/asterisk/sip.conf

    secret=verysecretpassword ; put a strong, unique password here instead
    ;permit= ; replace with your network settings
    secret=othersecretpassword ; put a strong, unique password here instead
    ;permit= ; replace with your network settings

    We have commented deny/permit lines with ";" character coz we would like to allow users to access server from anywhere. BUT! Asterisks wiki told us:

    Be Serious About Account Security We can't stress enough how important it is for you to pick a strong password for all accounts on Asterisk, and to only allow access from trusted networks. Unfortunately, we've found many instances of people exposing their Asterisk to the internet at large with easily-guessable passwords, or no passwords at all. You could be at risk of toll fraud, scams, and other malicious behavior. For more information on Asterisk security and how you can protect yourself, check out http://www.asterisk.org/security/webinar/. (c)https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts

    go to the Asterisk command-line interface. with simple command


    make sip reload with next command:

    sip reload

    Create Dialplan Extensions. add these lines into /etc/asterisk/extensions.conf


    The extension numbered 6001 which attempts to ring user1 phone for twenty seconds, and an extension 6002 which attempts to rings user2 phone for twenty seconds.

    make dialplan reload in the Asterisk command line interface

     dialplan reload

    You can verify that Asterisk successfully read the configuration file by typing dialplan show users at the CLI.

    dialplan show users

    You can now try to make test call using for example ExpressTalk software for windows (http://www.nch.com.au/talk/index.html)

    As another option you can configure your users in /etc/asterisk/users.conf Here is the example with NAT enabled, codecs and other options just for example. Simply add these lines to users.conf: adding 1 user

     fullname = test1
     registersip = no
     host = dynamic
     callgroup = 1
     mailbox = 6100
     call-limit = 100
     type = peer
     username = 6100
     transfer = yes
     callcounter = yes
     context = DLPN_DialPlan1
     cid_number = 6100
     hasvoicemail = no
     vmsecret =
     email =
     threewaycalling = no
     hasdirectory = yes
     callwaiting = no
     hasmanager = no
     hasagent = no
     hassip = yes
     hasiax = yes
     secret = P@ssword1
     nat = yes
     canreinvite = no
     dtmfmode = rfc2833
     insecure = no       
     pickupgroup = 1
     requirecalltoken = yes
     macaddress = 6100
     autoprov = yes
     label = 6100
     linenumber = 1        
     LINEKEYS = 1    
     disallow = all 
     allow = ulaw,g729,alaw,gsm

    if you want you can setup web gui for asteriks management regarding this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+GUI#AsteriskGUI-Download Also sometimes it's necessary to fix permissions for make GUI work:

    sudo chown -R asterisk:asterisk /var/lib/asterisk/static-http/

    Adding security

    You should download Asterisk source code if you haven't download it before from http://www.asterisk.org/downloads. Unzip it to some folder on your asterisk server.

    tar -xvzf asterisk-11-current.tar.gz 

    change dir to unzipped folder and run next command:

    contrib/scripts/./ast_tls_cert -C pbx.privatecompany.com -O "privatecompany" -d /etc/asterisk/keys

    You'll be asked to enter a pass phrase for /etc/asterisk/keys/ca.key NOTE if you got hostname related error pls make next : http://wiki.vpsget.com/index.php/Set_hostname

    Generate a client certificate for our SIP device.

    contrib/scripts/./ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C pbx.privatecompany.com -O "privatecompany" -d /etc/asterisk/keys -o sipuser1

    you also will be prompted to enter passphrase for keys Create cert for users. after check that all key files should be in the key directory:


    Configure Asterisk to use TLS: add/edit corresponding lines in sip.conf

    tlsclientmethod=tlsv1 ;none of the others seem to work with Blink as the client

    Next, you'll need to configure a SIP peer within Asterisk to use TLS as a transport type. add the line to your user/sip conf (it placed in sip.conf or in users.conf)

    port=5061 # not neccessary but it will force use tls

    Now you should copy keys from server to your client (pc or phone)

    Now you should configure your sip client to use tls via port 5061.

    How to make it depends from client you using.

    We will add soon manual for android CSip client

    TLS only encrypt extensions on the PBX we're dialing. We need to encrypt voice data with SRTP

    Asteriks wiki told us: SRTP support is provided by libsrtp. libsrtp has to be installed on the machine before Asterisk is compiled, otherwise you're going to see something like: [Jan 24 09:29:16] ERROR[10167]: chan_sip.c:27987 setup_srtp: No SRTP module loaded, can't setup SRTP session. But we installed it at first steps in our man. so don't care.

    Anyway if you got this in "asterisk -r" CLI during trying to make call do the next: install libsrtp (and the development header, and then reinstall Asterisk

    go to you asterisk source code directory and run next commands:

    make install

    If you're getting errors during ./configure is running make sure you have these packages installed:

    yum install gcc-c++ libtermcap-devel libxml2* sqlite-devel

    Add the next line to your user config (in sip.conf or in users.conf)


    Also better to force only one codec use:

    disallow = all
    ;allow = ulaw,g729,gsm <--this line is commented!
    allow = g722

    You can also restart asterisk service for sure.


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