The new parameter PHONENUMBER_TO_TG defines the comma-separated assignment of incoming SIP connections to a respective talkgroup. The format is 'PHONENUMBER_TO_TG=Sipnumber1:TG#1,Sipnumber2:TG#2,...'. Look into man page for further information.

src/doc/man/svxlink.conf.5

@@ -870,6 +870,16 @@ a compressor with a very steep compression ratio like 10:1. The limiter is
 used to help keeping down the audio level from Sip network for overmodulating
 stations. Try values from -6 to -12 if the incoming Sip audio is to high. Set
 to 0 to deactivate it.
+.TP
+.B PHONENUMBER_TO_TG
+Comma-separated list of assignments of phonenumber(full||substring):Talkgroup.
+When a phone call is received, the call is assigned to the defined TG according
+to the recognized number. This can be used to prevent the transfer of an SIP
+call to a network or to carry out a country-specific assignment. The number
+format transmitted must match the configuration (+49 != 0049). Whole SIP
+numbers as well as parts of phone numbers can be configured, with the
+evaluation taking place on the left-hand side.
+Example: PHONENUMBER_TO_TG=0049:262,0034:214,0043:232,0049123454543:991
 .
 .SS QSO Recorder Section
 .

src/svxlink/svxlink/SipLogic.cpp

@@ -56,6 +56,7 @@ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 #include <AsyncAudioCompressor.h>
 #include <AsyncAudioAmp.h>
 #include <AsyncAudioFilter.h>
+#include <common.h>
 
 
 /****************************************************************************
@@ -88,7 +89,7 @@ using namespace pj;
  *
  ****************************************************************************/
 #define DEFAULT_SIPLIMITER_THRESH  -1.0
-#define PJSIP_VERSION "30052022"
+#define PJSIP_VERSION "03062022"
 
 
 /****************************************************************************
@@ -532,6 +533,30 @@ bool SipLogic::initialize(void)
     return false;
   }
 
+  // set Tg's depending on the incoming sip phone number
+  std::string phonenumbers_to_tg;
+  if (cfg().getValue(name(), "PHONENUMBERS_TO_TG", phonenumbers_to_tg))
+  {
+    vector<string> nrlist;
+    SvxLink::splitStr(nrlist, phonenumbers_to_tg, ",");
+    for (vector<string>::const_iterator nr_it = nrlist.begin();
+       nr_it != nrlist.end(); nr_it++)
+    {
+      size_t pos;
+      if ((pos = (*nr_it).find(':')) != string::npos)
+      {
+        std::string r = (*nr_it).substr(0, pos);
+        uint32_t t = atoi((*nr_it).substr(pos+1, (*nr_it).length()-pos).c_str());
+        phoneNrTgVec[r] = t;
+      }
+      else
+      {
+        cout << "+++ WARNING: Wrong format in param " << name()
+             << "/PHONENUMBERS_TO_TG, ignoring." << endl;
+      }
+    }
+  }
+
    // create SipEndpoint - init library
   try {
     pj::EpConfig ep_cfg;
@@ -886,6 +911,17 @@ void SipLogic::onIncomingCall(sip::_Account *acc, pj::OnIncomingCallParam &iprm)
 
   std::string caller = getCallerNumber(ci.remoteUri);
 
+  for (std::map<std::string, uint32_t>::const_iterator it = phoneNrTgVec.begin();
+      it != phoneNrTgVec.end(); it++)
+  {
+    size_t pos;
+    if ((pos = caller.find(it->first)) == 0)
+    {
+      uint32_t tg = it->second;
+      setReceivedTg(tg);
+    }
+  }
+
   stringstream ss;
   ss << "ringing \"" << caller << "\"";
   processLogicEvent(ss.str());
@@ -1086,6 +1122,8 @@ void SipLogic::onCallState(sip::_Call *call, pj::OnCallStateParam &prm)
        << ci.totalDuration.sec << "."
        << ci.totalDuration.msec;
 
+    setReceivedTg(0);
+
     // if no one is connected anymore, call out to autoconnect party
     if (calls.empty() && m_autoconnect.length() > 0)
     {

src/svxlink/svxlink/SipLogic.h

@@ -203,6 +203,7 @@ class SipLogic : public LogicBase
     EventHandler              *sip_event_handler;
     MsgHandler                *sip_msg_handler;
     Async::AudioSelector      *sipselector;
+    std::map<std::string, uint32_t> phoneNrTgVec;
 
     SipLogic(const SipLogic&);
     SipLogic& operator=(const SipLogic&);

src/svxlink/svxlink/svxlink.conf.in

@@ -135,6 +135,7 @@ SIP_CTRL_PTY=/tmp/sip_pty
 EVENT_HANDLER=@SVX_SHARE_INSTALL_DIR@/events.tcl
 SIP_PREAMP=3
 SIP_LIMITER_THRESH=-1.0
+PHONENUMBER_TO_TG=0049:262,0034:232,0041:222,017612345678:9999
 
 [LinkToR4]
 CONNECT_LOGICS=RepeaterLogic:94:SK3AB,SimplexLogic:92:SK3CD