fix outbound sip uri To header and Request Line are not same

lib/call-session.js

@@ -348,9 +348,10 @@ class CallSession extends Emitter {
             this.logger.info({err}, 'Error retrieving iip key');
           }
         }
+        // INVITE request line and To header should be the same.
+        hdrs = {...hdrs, 'To': uri};
         if (gw) {
           this.logger.info({gw}, `sending INVITE to ${uri} via carrier ${gw.name}`);
-          hdrs = {...hdrs, 'To': uri};
           if (gw.diversion) {
             let div = gw.diversion;
             if (div.startsWith('+')) {

test/scenarios/uac-pcap-sip-routing-success.xml

@@ -0,0 +1,91 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAC with media">
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:16173333456@172.39.0.20 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:16173333456@172.39.0.20>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      X-Account-Sid: ed649e33-e771-403a-8c99-1780eabbc803
+      X-Call-Sid: fff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: sip
+      Subject: uac-pcap-carrier-success
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[local_ip_type] [local_ip]
+      t=0 0
+      m=audio [auto_media_port] RTP/AVP 8 101
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:101 telephone-event/8000
+      a=fmtp:101 0-11,16
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200" rtd="true" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:sip:+16173333456@127.0.0.1 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Subject: uac-pcap-carrier-success
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
+  <nop>
+    <action>
+      <exec play_pcap_audio="pcap/g711a.pcap"/>
+    </action>
+  </nop>
+
+  <!-- Pause briefly      -->
+  <pause milliseconds="2000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:sip:+16173333456@127.0.0.1 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Max-Forwards: 70
+      Subject: uac-pcap-carrier-success
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+</scenario>

test/sip-tests.js

@@ -44,6 +44,10 @@ test('sbc-outbound tests', async(t) => {
     await sippUac('uac-pcap-carrier-success.xml');
     t.pass('successfully completed outbound call to sip trunk');
 
+    /* call to Sip URI with no lcr configured */
+    await sippUac('uac-pcap-sip-routing-success.xml');
+    t.pass('successfully completed outbound call to sip routing trunk');
+
     /* call to PSTN with no lcr configured */
     await sippUac('uac-pcap-inbound-carrier-success.xml');
     t.pass('successfully completed outbound call to sip trunk');