Release 4.6

config/golang.m4

@@ -0,0 +1,16 @@
+##
+## AC macros for using go lang
+##
+
+AC_DEFUN([CHECK_GOLANG],[
+	AC_ARG_VAR(GOLANG, [ Go compiler full path ])
+
+	AC_PATH_PROG([GOLANG],[go],[no])
+	if test x$GOLANG != xno; then
+		found_golang="yes";
+	fi
+	AS_VAR_SET_IF([found_golang],
+	[],
+	[AC_MSG_ERROR([Cannot find golang installed. Install go and set GOROOT var])]
+	)
+])

sipXbridge/src/main/java/org/sipfoundry/sipxbridge/SipUtilities.java

@@ -1042,16 +1042,35 @@ static Set<Integer> getNonTelephoneEventMediaFormats(
 			for (Iterator it = mediaDescriptions.iterator(); it.hasNext();) {
 				MediaDescription mediaDescription = (MediaDescription) it
 						.next();
+				int telephoneEvent = -1;
 				if (mediaDescription.getMedia().getMediaType().equals("audio")) {
-					Vector formats = mediaDescription.getMedia()
-							.getMediaFormats(true);
-					for (Iterator it1 = formats.iterator(); it1.hasNext();) {
-						Object format = it1.next();
-						int fmt = new Integer(format.toString());
-						if (fmt != 100 && fmt != 101) {
-							retval.add(fmt);
+				    Vector attributes = mediaDescription.getAttributes(true);
+
+				    for (Iterator it1 = attributes.iterator(); it1.hasNext();) {
+					Attribute attr = (Attribute) it1.next();
+
+					if (attr.getName().equalsIgnoreCase("rtpmap") || attr.getName().equalsIgnoreCase("fmtp")) {
+					    String attribute = attr.getValue();
+					    String[] attrs = attribute.split(" ");
+					    if (!attrs[1].startsWith("telephone-event")) {
+						try{
+						    telephoneEvent = Integer.parseInt(attrs[0]);
+						} catch ( NumberFormatException ex) {
+						    logger.warn("could not parse RTP Map " + attrs[0]);
+						    continue;
 						}
+					    }
 					}
+				    }
+				    Vector formats = mediaDescription.getMedia()
+							.getMediaFormats(true);
+				    for (Iterator it1 = formats.iterator(); it1.hasNext();) {
+					Object format = it1.next();
+					int fmt = new Integer(format.toString());
+					if (fmt != telephoneEvent) {
+					    retval.add(fmt);
+					}
+				    }
 				}
 			}
 			return retval;
@@ -1127,27 +1146,8 @@ static SessionDescription cleanSessionDescription(
 				Vector formats = mediaDescription.getMedia().getMediaFormats(
 						true);
 				String mediaType = mediaDescription.getMedia().getMediaType();
-
+				int telephoneEvent = -1;
 				if ( mediaType.equals("audio")) {
-					for (Iterator it1 = formats.iterator(); it1.hasNext();) {
-						Object format = it1.next();
-						try {
-							Integer fmt = new Integer(format.toString());
-							if (!codecs.contains(fmt)) {
-								/*
-								 * Preserve the telephone event lines -- this upsets
-								 * some ITSPs otherwise. AT&T Hack.
-								 */
-								if (fmt != 100 && fmt != 101) {
-									it1.remove();
-								}
-							}
-						} catch (NumberFormatException ex) {
-							logger.warn("Could not parse media format " + format);
-							continue;
-						}
-					}
-
 					Vector attributes = mediaDescription.getAttributes(true);
 					for (Iterator it1 = attributes.iterator(); it1.hasNext();) {
 						Attribute attr = (Attribute) it1.next();
@@ -1161,17 +1161,18 @@ static SessionDescription cleanSessionDescription(
 							String[] attrs = attribute.split(" ");
 							try {
 								int rtpMapCodec = Integer.parseInt(attrs[0]);
-
-								if (!codecs.contains(rtpMapCodec)) {
-									/*
-									 * Preserve the telephone event lines -- this upsets
-									 * some ITSPs otherwise. AT&T Hack.
-									 */
-									if (rtpMapCodec != 100 && rtpMapCodec != 101) {
+								/*
+								 * Preserve the telephone event lines -- this upsets
+								 * some ITSPs otherwise. AT&T Hack.
+								 */
+								if (!codecs.contains(rtpMapCodec) && rtpMapCodec!=telephoneEvent) {
+								        if (!attrs[1].startsWith("telephone-event")) {
 									    if ( logger.isDebugEnabled()) {
 									        logger.debug("codecs does not contain " + rtpMapCodec + " removing it ");
 									    }
 										it1.remove();
+									} else {
+									    telephoneEvent=rtpMapCodec;
 									}
 								}
 							} catch ( NumberFormatException ex) {
@@ -1187,6 +1188,23 @@ static SessionDescription cleanSessionDescription(
 						}
 
 					}
+					for (Iterator it1 = formats.iterator(); it1.hasNext();) {
+						Object format = it1.next();
+						try {
+							Integer fmt = new Integer(format.toString());
+							/*
+							 * Preserve the telephone event lines -- this upsets
+							 * some ITSPs otherwise. AT&T Hack.
+							 */
+							if (!codecs.contains(fmt) && telephoneEvent!=fmt) {
+							    it1.remove();
+							}
+						} catch (NumberFormatException ex) {
+							logger.warn("Could not parse media format " + format);
+							continue;
+						}
+					}
+
 				}
 
 			}

sipXcdr/bin/sipxcdr.in

@@ -137,8 +137,9 @@ EOF
 # special patch that upgrades DB version to the latest version - only use once
 VersionPatch="upgrade_version"
 
-UpgradePatches7=""
-UpgradePatches6="cdrremote $VersionPatch"
+UpgradePatches8=""
+UpgradePatches7="gateway $VersionPatch"
+UpgradePatches6="cdrremote $UpgradePatches7"
 UpgradePatches5="transaction $UpgradePatches6"
 UpgradePatches4="reference contact calldirection transaction $UpgradePatches5"
 UpgradePatches3="index_on_time $UpgradePatches4"

sipXcdr/etc/Makefile.am

@@ -10,7 +10,8 @@ dist_sql_DATA = \
 	database/contact.sql \
 	database/calldirection.sql \
 	database/cdrremote.sql \
-	database/transaction.sql
+	database/transaction.sql \
+	database/gateway.sql
 
 cfinputsdir = @SIPX_CFINPUTS@/plugin.d
 dist_cfinputs_DATA = \

sipXcdr/etc/database/gateway.sql

@@ -0,0 +1,2 @@
+alter table cdrs add column called_number text ;
+alter table cdrs add column gateway int ;

sipXcdr/lib/cdr.rb

@@ -153,6 +153,7 @@ def established?
   private
   def get_leg(cse)
     id = CallLeg.leg_id(cse)
+    puts cse.request_uri,"->", id
     @legs[id] ||= CallLeg.new(id, @log)
   end
 end
@@ -207,7 +208,10 @@ def initialize(call_id, log=nil)
     @failure_status = nil
     @failure_reason = nil
     @call_direction = nil
-
+    @called_number = nil
+    @gateway = nil
+    @contact_info = {}
+
     @got_original = false
     @log = log
     @legs = CallLegs.new(@log)
@@ -223,7 +227,8 @@ def initialize(call_id, log=nil)
   FIELDS = [:call_id, :from_tag, :to_tag, :caller_aor, :callee_aor, 
   :start_time, :connect_time, :end_time,    
   :termination, :failure_status, :failure_reason, 
-  :call_direction, :reference, :caller_contact, :callee_contact, :caller_internal, :callee_route]
+  :call_direction, :reference, :caller_contact, :callee_contact, :caller_internal, :callee_route,
+  :called_number, :gateway ]
 
   attr_accessor(*FIELDS)
 
@@ -301,8 +306,14 @@ def accept_call_request(cse)
     original = !cse.to_tag
     @caller_internal = cse.caller_internal 
     @via_count = cse.via_count if (!@via_count || cse.via_count < @via_count) 
-    @branch_id = cse.branch_id if (!@branch_id || cse.via_count <= @via_count) 
+    @branch_id = cse.branch_id if (!@branch_id || cse.via_count <= @via_count)
     # bailout if we already have original request and this one is not
+
+    if cse.request_uri.include? "sipxecs-lineid"
+      called_number = Utils.contact_without_params(cse.request_uri)
+      gateway = Integer(/.*sipxecs-lineid=(\d+).*/.match(cse.request_uri)[1])
+      @contact_info[cse.branch_id] = [called_number, gateway]
+    end
     return if(@got_original && !original)
 
     # continue if we are original or if we are older 
@@ -323,6 +334,10 @@ def accept_call_request(cse)
   end
 
   def accept_call_setup(cse)
+    if @contact_info.has_key? cse.branch_id
+      @called_number = @contact_info[cse.branch_id][0]
+      @gateway = @contact_info[cse.branch_id][1]
+    end
     if !@start_time
         # probably a case where we've missed the request.  Setup necessary
         # info as if a request was seen.
@@ -370,7 +385,7 @@ def apply_leg(leg)
     end
     @failure_reason = leg.failure_reason
     @failure_status = leg.failure_status
-    @callee_contact = leg.callee_contact  
+    @callee_contact = leg.callee_contact
   end
 
   # Map termination codes to human-readable strings

sipXconfig/etc/ftp.cf

@@ -89,7 +89,8 @@ pam_service_name=vsftpd
 listen=YES
 chroot_local_user=YES
 port_enable=YES
-log_ftp_protocol=YES";
+log_ftp_protocol=YES
+user_config_dir=/etc/vsftpd/user_conf";
     "$(sipx.SIPX_CFDATA)/$(sipx.location_id)/vsftp.config.part"
       insert_type => "file";
 }

sipXconfig/etc/sipxpbx/commserver/forwardingrules.vm

@@ -22,32 +22,28 @@
     <routeFrom>$route</routeFrom>
 #end	    
     <methodMatch>
-      <methodPattern>SUBSCRIBE</methodPattern>
-#if( ${statusAddress} )
-
-      <!-- Message waiting indication events -->
-      <fieldMatch fieldName="Event">
-        <fieldPattern>message-summary.*</fieldPattern>
-        <!--this goes to the status server-->
-        <routeTo>&lt;${statusAddress.stripProtocol()};transport=tcp&gt;</routeTo>
-      </fieldMatch>
-
+#set($method = $locals.get("subscribe"))
+#set($methodDefault = $method.getDefRoute())
+#set($methodName = $method.getName())
+      <methodPattern>$methodName</methodPattern>
+#foreach ($field in $method.getFields())
+#if($field.get("name"))
+
+        <fieldMatch fieldName=$field.get("name")>
+          <fieldPattern>$field.get("pattern")</fieldPattern>      
+          <routeTo>$field.get("routeTo")</routeTo>
+        </fieldMatch>
+#end
 #end
-      <!-- "reg" event subscriptions. -->
-      <fieldMatch fieldName="Event">
-        <fieldPattern>reg</fieldPattern>
-        <routeTo>&lt;${regEventAddress.stripProtocol()};transport=tcp&gt;</routeTo>
-      </fieldMatch>
 
-      <!-- All other SUBSCRIBE requests go to the SIP registry service -->
-      <routeTo>&lt;${regAddress.stripProtocol()};transport=tcp;x-sipx-routetoreg&gt;</routeTo>
+	<routeTo>$methodDefault</routeTo>
 
     </methodMatch>
-
     <!-- All other requests go to the SIP registry service -->
     <routeTo>&lt;${regAddress.stripProtocol()};transport=tcp;x-sipx-routetoreg&gt;</routeTo>
   </route>
 
+
 #if(${sbc.enabled})
 #if(${exportLocalIpAddress})
     <route mappingType='local ip address'>
@@ -66,6 +62,7 @@
       <routeTo authRequired="true"/>
     </route>
 #end
+
 #foreach($auxSbc in $auxSbcs)
 #if($auxSbc.enabled)
   <route mappingType='auxiliary SBC'>
@@ -86,6 +83,7 @@
 #end
 #end
 
+
   <route mappingType="external destinations">
     <description>Any foreign domain - route via session border.</description>
 

sipXconfig/etc/sipxpbx/dhcp/dhcp.properties

@@ -12,6 +12,8 @@ dhcpd-config.plcm_https_provisioning.label=Polycom HTTPS provisioning
 dhcpd-config.plcm_https_provisioning.description=Enable Polycom phones and other devices that utilize DHCP option 160 to use HTTPS provisioning. Please note that the web SSL certificate MUST signed by a well know Cerificate Authority such as Verisign. 
 dhcpd-config.bria_https_provisioning.label=CounterPath Bria HTTPS provisioning
 dhcpd-config.bria_https_provisioning.description=Enable CounterPath Bria clients and other devices that utilize DHCP option 120 to use HTTPS provisioning. Please note that the web SSL certificate MUST signed by a well know Cerificate such as Verisign.
+dhcpd-config.snom_statichttp_provisioning.label=Adding Snom specific class
+dhcpd-config.snom_statichttp_provisioning.description=Enable definition of snom class that provides url.
 dhcpd-config.default_lease_time.label=Default lease time
 dhcpd-config.default_lease_time.description=Length in seconds that will be assigned to a lease if the client requesting the lease does not ask for a specific expiration time.
 dhcpd-config.max_lease_time.label=Maximum lease time

sipXconfig/etc/sipxpbx/dhcp/dhcp.xml

@@ -2,6 +2,7 @@
 <!DOCTYPE model PUBLIC "-//SIPFoundry//sipXconfig//Model specification 2.0//EN"
   "http://www.sipfoundry.org/sipXconfig/dtd/setting_2_0.dtd">
 <model>
+
   <group name="dhcpd-config">
     <setting name="subnet" >
       <type><ipaddr/></type>
@@ -39,6 +40,12 @@
       </type>
       <value>0</value>
     </setting>
+    <setting name="snom_statichttp_provisioning">
+      <type>
+        <boolean />
+      </type>
+      <value>0</value>
+    </setting>
  </group>
   <group name="dhcpd-unmanaged">
     <setting name="unmanaged">

sipXconfig/etc/sipxpbx/sipxcallresolver/sipxcallresolver.properties

@@ -34,4 +34,11 @@ callresolver.SIP_CALLRESOLVER_CSE_QUEUE_SIZE.label=CSE Queue size
 callresolver.SIP_CALLRESOLVER_CDR_QUEUE_SIZE.label=CDR Queue size
 callresolver.SIP_CALLRESOLVER_MAX_CALL_LEN.label=Maximum call length
 callresolver.SIP_CALLRESOLVER_MAX_RINGING_CALL_LEN.label=Maximum Ringing call length
-callresolver.SIP_CALLRESOLVER_MIN_CLEANUP_INTERVAL.label=Minimum cleanup interval
+callresolver.SIP_CALLRESOLVER_MIN_CLEANUP_INTERVAL.label=Minimum cleanup interval
+callresolver.SIP_CALLRESOLVER_PRIVACY.label=Privacy Extension
+callresolver.SIP_CALLRESOLVER_PRIVACY.description=When enabled hides last digits of phone numbers into cdrs to enhance privacy
+callresolver.SIP_CALLRESOLVER_PRIVACY_LENGTH.label=Minimum length of extension that must be masquerade
+callresolver.SIP_CALLRESOLVER_PRIVACY_LENGTH.description=This value indicate the minimum length of extension that should be covered with *
+callresolver.SIP_CALLRESOLVER_PRIVACY_EXCLUDE.label=Prefix excluded from masquerading
+callresolver.SIP_CALLRESOLVER_PRIVACY_EXCLUDE.description=This helps to exclude some prefix to be masquerade for extension with length equal to minimum. 
+

sipXconfig/etc/sipxpbx/sipxcallresolver/sipxcallresolver.xml

@@ -116,5 +116,21 @@
       </type>
       <value />
     </setting>
+    <setting name="SIP_CALLRESOLVER_PRIVACY" advanced="yes">
+      <type refid="enablestate" />
+      <value>DISABLE</value>
+    </setting>
+    <setting name="SIP_CALLRESOLVER_PRIVACY_LENGTH" advanced="yes">
+      <type>
+        <integer min="4" max="25" />
+      </type>
+      <value>5</value>
+    </setting>
+    <setting name="SIP_CALLRESOLVER_PRIVACY_EXCLUDE" advanced="yes">
+      <type>
+        <string />
+      </type>
+      <value></value>
+    </setting>
   </group>
 </model>

sipXconfig/etc/sipxpbx/sipxproxy/sipXproxy-config.vm

@@ -11,6 +11,9 @@ SIPX_PROXY_STALE_TCP_TIMEOUT :
 SIPX_PROXY_TLS_PORT : $!{proxyService.secureSipPort}
 SIPX_PROXY_TCP_PORT : $!{proxyService.sipTCPPort}
 SIPX_PROXY_UDP_PORT : $!{proxyService.sipUDPPort}
+SIPX_PROXY_UDP_RESEND : $!{proxyService.sipUDPResend}
+SIPX_PROXY_TCP_RESEND : $!{proxyService.sipTCPResend}
+
 SIPX_PROXY_CALL_STATE :
 SIPX_PROXY_CALL_STATE_LOG :
 SIPX_PROXY_CALL_STATE_DB : $callResolverSettings.getSetting('CALLRESOLVER_CALL_STATE_DB').Value

sipXconfig/etc/sipxpbx/sipxproxy/sipxproxy.properties

@@ -33,6 +33,8 @@ alert-info.ON_EXISTING.description=Phones can be configured to send their own Al
 type.replaceOrKeep.0=Leave it as it is
 type.replaceOrKeep.1=Replace it
 
+proxy-configuration.SIPX_PROXY_UDP_RESEND.label=Max resend attempts for UDP
+proxy-configuration.SIPX_PROXY_TCP_RESEND.label=Max resend attempts for TCP
 proxy-configuration.SIPX_PROXY_LOG_LEVEL.label=Log Level
 proxy-configuration.SIPX_PROXY_TCP_PORT.label=TCP Port
 proxy-configuration.SIPX_PROXY_TCP_PORT.description=