Sample 1
Topology
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// Some comments here
listeners :
{
interfaces = (
{
default = true;
ip-address = "10.0.40.5";
external-address = "10.0.40.5";
tcp-enabled = false;
sip-port = 5060;
},
{
ip-address = "192.168.2.2";
external-address = "192.168.2.2";
tcp-enabled = false;
sip-port = 5060;
} );
/****************************************************************************
* The default interface address used to send out requests if not explicitly*
* set using the java script sendInterfaceAddress() function. This is also *
* used by Karoo Bridge command line utilities like the --reset-log-level *
* to send the SIP RPC request to the running Karoo Bridge instance. *
****************************************************************************/
default-interface-address = "10.0.40.5";
/****************************************************************************
* The default interface port used to send out requests if not explicitly *
* set using the java script sendInterfacePort() function *
****************************************************************************/
default-interface-port = 5060;
/****************************************************************************
* The low value marker for the TCP client port range *
****************************************************************************/
sip-tcp-port-base = 10000;
/****************************************************************************
* The high value marker for the TCP client port range *
****************************************************************************/
sip-tcp-port-max = 15000;
/****************************************************************************
* The low value marker for the RTP proxy port range *
****************************************************************************/
rtp-proxy-port-base = 30000;
/****************************************************************************
* The high value marker for the RTP proxy port range *
****************************************************************************/
rtp-proxy-port-max = 60000;
/****************************************************************************
* The read timeout for RTP Proxy before it decides to automatically *
* tear down the RTP channel. This is expressed in seconds. *
****************************************************************************/
rtp-proxy-read-timeout = 600;
/****************************************************************************
* The number of worker threads for the RTP proxy. Take note that Karoo *
* Bridge uses async IO and the number here does not limit the number of RTP*
* channels that can be supported. This threadpool will be shared equally *
* by all RTP channels. 30 threads has been tested to be enough to cater *
* to 1000 RTP proxy sessions *
****************************************************************************/
rtp-proxy-transport-thread-count = 30;
/***************************************************************************
* Karoo has a built-in STUN server. STUN would only work if the server *
* Karoo Bridge is installed in has two or more NICs directly routable from*
* the WAN. The addresses must be different. Virtual IP is also ok but is*
* not advised. If you don't want to use the default port you can include *
* a port with the address separated with a colon. *
* Example: stun-primary-ip = 10.0.40.5:3478 *
***************************************************************************/
stun-primary-ip = "";
stun-secondary-ip = "";
/***************************************************************************
* Karoo bridge uses libfreeswitch for sip trunking and transcoding *
* functions. These values set the listener port for the internal event *
* layer handler. Since freeswitch is configured independently, the sip *
* address used by freeswitch must be made known to Karoo as well *
***************************************************************************/
switch-event-listener-port = 9000;
switch-sip-listener-ip = "10.0.40.5";
switch-sip-listener-port = 5080;
switch-sip-context = "sofia/karoo";
/****************************************************************************
* The packet rate ratio allows the transport to detect a potential DoS *
* attack. It works by detecting the packet read rate per second as *
* designated by the upper limit. If the value of packet rate is 50/100, *
* the maximum packet rate before the SBC raises the alert level if a *
* potential denial of service attack is 100 packets per second. *
* When this happens, the transport layer checks if there is a particular *
* IP that is sending more than its allowable rate of 50 packets per second.*
* If the sender is violating the threshold, it will be banned for 1 hour *
* which is the third parameter of 3600 seconds. *
****************************************************************************/
packet-rate-ratio = "50/100/3600";
/*****************************************************************************
* One may statically define a list of known IP addresses or networks so that*
* they get immunity against the packet-rate-ratio algorithm. This would *
* normally contain the ip addresses of known traffic sources or destinations*
* like the local iPBX or trunk gateways. For Call Centers with predictive *
* or progressive dialers in the network, it would be wise to also white-list*
* those applications. *
*****************************************************************************/
packet-rate-white-list = (
/* {source-ip = "192.168.1.10";\}, */
{source-network = "1.1.1.1/4";\}
\);
\};
user-agent :
\{
/****************************************************************************
* Show your appreciation by not changing this value *
****************************************************************************/
user-agent-name = "OSS Karoo Bridge";
/****************************************************************************
* Karoo bridge uses the contact header to store some states about sip *
* session. It can either do thi by storing a unique id as the user info of*
* the contact-uri or by storing it as a contact paramameter. If your *
* switch/pbx supports paramaters in contact, setting the two parameters *
* below to true is the recommended value. *
****************************************************************************/
register-state-in-contact-params=false;
dialog-state-in-contact-params=true;
\};
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| xml | xml | <test> <another tag="attribute"/> </test>
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// Some comments here
public String getFoo()
{
return foo;
}
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function Route()
{
this.sendRegister(
"user", "myitsp.com", "passwd", "proxy.myitsp.com", 1800, 0);
this.sendRegister(
"user2", "myitsp.com", "passwd2", "proxy.myitsp.com", 1800, 0);
}
Route.prototype = new RouteProfile();
Route.prototype.isRoutable = function()
{
if (is_mysipdomain_routable(this))
return true;
else
return false;
}
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function is_mysipdomain_routable(profile)
{
var ifaceAddr = profile.sipMessage.getInterfaceAddress();
var ifacePort = profile.sipMessage.getInterfacePort();
var ds = profile.sipMessage.getRequestUriUser();
var domain = profile.sipMessage.getFromHost();
var ts = profile.sipMessage.getToUser();
if (is_my_pbx_routable(profile, ifaceAddr, ifacePort, ds, domain, ts))
return true;
else if (is_my_sip_trunk_routable(profile, ifaceAddr, ifacePort, ds, domain))
return true;
}
function is_my_sip_trunk_routable(profile, ifaceAddr, ifacePort, ds, domain)
{
if (typeof ds != "string")
return false;
//
// Check if the packet came in using the LAN interface
//
if ( ifaceAddr != sip_interface_address[1] ||
ifacePort != sip_interface_port[1])
return false;
//
// Route it to our ITSP
//
profile.setTargetDomain("myitsp.com");
profile.setTargetAddress("udp", "proxy.myitsp.com", "5060");
//
// Use the WAN interface to send it out
//
profile.setInterfaceAddress(sip_interface_address[0], sip_interface_port[0]);
//
// Bridge it using our account
//
var pattern=/\b((\d\d\d)|((1)\d\d\d))?(\d{7})\b/;
var pattern2=/\b9((\d\d\d)|((1)\d\d\d))?(\d{7})\b/;
var pattern_ac=/\b(1?(\d\d\d))\b/;
var pattern_am=/\b((\d\d\d)(\d\d\d)?)\b/;
if ( pattern.test(ds) == true )
{
profile.sipMessage.setRequestUriUser( "1" + (ds.replace(pattern, "$1").replace(pattern_ac, "$2") + "888").replace(pattern_am, "$2") + ds.replace(pattern, "$5") );
profile.sipMessage.setFromUser("user");
profile.bridge("user", "passwd");
}
else if ( pattern2.test(ds) == true )
{
profile.sipMessage.setRequestUriUser( "1" + (ds.replace(pattern2, "$1").replace(pattern_ac, "$2") + "888").replace(pattern_am, "$2") + ds.replace(pattern2, "$5") );
profile.sipMessage.setFromUser("user2");
profile.bridge("user2", "passwd2");
}
else return false;
return true;
}
function is_my_pbx_routable(profile, ifaceAddr, ifacePort, ds, domain, ts)
{
if (typeof domain != "string")
return false;
//
// Check if the packet came in using the WAN interface
//
if ( ifaceAddr != sip_interface_address[0] ||
ifacePort != sip_interface_port[0])
return false;
//
// Check if the domain is valid
//
if (domain != "mydomain.com")
return false;
//
// route it to our PBX
//
profile.setTargetDomain("mydomain.com");
profile.setTargetAddress("udp", "sipx.mydomain.com", "5060");
//
// Use the LAN interface to send it out
//
profile.setInterfaceAddress(sip_interface_address[1], sip_interface_port[1]);
// profile.setInterfaceAddress("192.168.2.2", "5060");
//
// Check if the source address is the ITSP. If yes, bridge it
//
if (profile.sipMessage.getSourceAddress() == "10.0.168.10")
{
if (ts == "1XXXXXXX181")
{
profile.sipMessage.setRequestUriUser("501");
profile.bridge("501", "sipxpasswd");
}
else if (ts == "1XXXXXXX139")
{
profile.sipMessage.setRequestUriUser("503");
profile.bridge("503", "sipxpasswd");
}
else if (ts == "1XXXXXXXX07")
{
profile.sipMessage.setRequestUriUser("504");
profile.bridge("504", "sipxpasswd");
}
else if (ts == "1XXXXXXXX82")
{
profile.sipMessage.setRequestUriUser("1XXXXXXXX82");
profile.bridge("514", "sipxpasswd");
}
else if (ts == "1XXXXXXXX67")
{
profile.sipMessage.setRequestUriUser("501");
profile.bridge("501", "sipxpasswd");
}
}
return true;
}
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