Hi all,I'm running FS 1.4.9 on a CentOS 6.5 machine with a single NIC. The machine is behind NAT, and I have a working VPN so that internal calls work. The local IP of the machine is 192.168.196.101, and the remote VPN subnet (no NAT - clear routing) is 192.168.41.0/24.I am using port 5080 for internal calls (auth-calls=true) and port 5060 for incoming calls from a SIP provider. The machine has an internal IP and port-forwarding handles public IP -> private IP.When I dial from a mobile phone to a number on the external SIP provider, I do get activity on the console:2014-10-10 10:52:20.655471 [NOTICE] switch_channel.c:1055 New Channel sofia/external/07951357XXX@87.238.73.181 [78cc73d0-e9d1-4d7d-9f22-f7adac8ba0c0]2014-10-10 10:52:21.259045 [NOTICE] switch_ivr.c:1844 Transfer sofia/external/07951357xxx@87.238.73.181 to XML[106@default]2014-10-10 10:52:21.259045 [WARNING] mod_dptools.c:1628 incoming call from [07951357XXX] to [106]2014-10-10 10:52:21.259045 [NOTICE] mod_sofia.c:2237 Pre-Answer sofia/external/07951357XXX@87.238.73.181!2014-10-10 10:52:21.259045 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:106@192.168.41.198:5060 [a7aaa63b-fd85-4b1f-8855-180a218a388e]At this point there is a pause and a timeout after 30 seconds. There is a phone registered to extension '106' listening at that IP address on port 5060:freeswitch@internal> sofia status profile internal regRegistrations:=================================================================================================Call-ID: 1572302163@192.168.41.198Contact: "" <sip:106@192.168.41.198:5060>Agent: qutecom/rev-g-trunkStatus: Registered(UDP)(unknown) EXP(2014-10-10 10:32:04) EXPSECS(2796)Host: freeswitchIP: 192.168.41.198Port: 5060Auth-User: 106Auth-Realm: freeswitch.bashton.euMWI-Account: 106@freeswitch.bashton.euTotal items returned: 1=================================================================================================freeswitch@internal> sofia status profile external=================================================================================================Name externalDomain Name N/AAuto-NAT falseDBName sofia_reg_externalPres HostsDialplan XMLContext publicChallenge Realm auto_toRTP-IP 192.168.196.101SIP-IP 192.168.196.101BIND-URL sip:mod_sofia@192.168.196.101:5060;transport=udp,tcpHOLD-MUSIC N/AOUTBOUND-PROXY N/ACODECS IN PCMU,PCMACODECS OUTTEL-EVENT 101DTMF-MODE rfc2833CNG 13SESSION-TO 0MAX-DIALOG 0NOMEDIA falseLATE-NEG falsePROXY-MEDIA falseZRTP-PASSTHRU falseAGGRESSIVENAT falseCALLS-IN 2FAILED-CALLS-IN 1CALLS-OUT 0FAILED-CALLS-OUT 0REGISTRATIONS 0freeswitch@internal> sofia status profile internal=================================================================================================Name internalDomain Name N/AAuto-NAT falseDBName sofia_reg_internalPres HostsDialplan XMLContext defaultChallenge Realm auto_toRTP-IP 192.168.196.101Ext-RTP-IP 192.168.196.101SIP-IP 192.168.196.101Ext-SIP-IP 192.168.196.101BIND-URL sip:mod_sofia@192.168.196.101:5080;maddr=192.168.196.101;transport=udpHOLD-MUSIC N/AOUTBOUND-PROXY N/ACODECS IN PCMU,PCMACODECS OUT PCMU,PCMATEL-EVENT 101DTMF-MODE rfc2833CNG 13SESSION-TO 0MAX-DIALOG 0NOMEDIA falseLATE-NEG falsePROXY-MEDIA falseZRTP-PASSTHRU falseAGGRESSIVENAT falseCALLS-IN 0FAILED-CALLS-IN 0CALLS-OUT 2FAILED-CALLS-OUT 2REGISTRATIONS 1freeswitch@internal> sofia statusName Type Data State==================================================================================================================================================================================================2 profiles 0 aliasesThe SIP client is definitely listening:gdh@gdh-e7440:~$ netstat -anp | grep 5060tcp 0 0 192.168.41.198:5060 0.0.0.0:* LISTEN 17566/qutecomudp 0 0 192.168.41.198:5060 0.0.0.0:* 17566/qutecomAnd I can see that when the call comes in, there is traffic received by the SIP client (tcpdump + wireshark):INVITE sip:106@192.168.41.198:5060 SIP/2.0Via: SIP/2.0/UDP 192.168.196.101:5080;rport;branch=z9hG4bKX71j6K414rQNaMax-Forwards: 67From: "None" <sip:07951357XXX@freeswitch.bashton.eu>;tag=rBeB1Nv1BampmCall-ID: f850b9a6-cb05-1232-7693-02b891c191f6CSeq: 66131306 INVITEContact: <sip:mod_sofia@192.168.196.101:5080>User-Agent: FreeSWITCH-mod_sofia/1.4.9~64bitAllow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFYSupported: timer, path, replacesAllow-Events: talk, hold, conference, referContent-Type: application/sdpContent-Disposition: sessionContent-Length: 253X-CALLINFO: cdr=5437AC53=94F6B14;X-FS-Support: update_display,send_infoRemote-Party-ID: "None" <sip:07951357XXX@freeswitch.bashton.eu>;party=calling;screen=yes;privacy=offThe incoming call is handled in the public context with:<action application="transfer" data="106 XML default"/>and in extension 106 in the default context, there's this:<action application="bridge" data="user/$1"/>I have tried both qutecom and linphone with the same behaviour. Any ideas?Cheers,Gavin.
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