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 reg Registrations: ================================================================================================= Call-ID: 1572302163@192.168.41.198 User: 106@freeswitch.bashton.eu Contact: "" <sip:106@192.168.41.198:5060> Agent: qutecom/rev-g-trunk Status: Registered(UDP)(unknown) EXP(2014-10-10 10:32:04) EXPSECS(2796) Host: freeswitch IP: 192.168.41.198 Port: 5060 Auth-User: 106 Auth-Realm: freeswitch.bashton.eu MWI-Account: 106@freeswitch.bashton.eu Total items returned: 1 ================================================================================================= freeswitch@internal> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.196.101 SIP-IP 192.168.196.101 URL sip:mod_sofia@192.168.196.101:5060 BIND-URL sip:mod_sofia@192.168.196.101:5060;transport=udp,tcp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN PCMU,PCMA CODECS OUT TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 2 FAILED-CALLS-IN 1 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 freeswitch@internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts Dialplan XML Context default Challenge Realm auto_to RTP-IP 192.168.196.101 Ext-RTP-IP 192.168.196.101 SIP-IP 192.168.196.101 Ext-SIP-IP 192.168.196.101 URL sip:mod_sofia@192.168.196.101:5080 BIND-URL sip:mod_sofia@192.168.196.101:5080 ;maddr=192.168.196.101;transport=udp HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN PCMU,PCMA CODECS OUT PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 REGISTRATIONS 1 freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@192.168.196.101:5060 RUNNING (0) external::magrathea gateway sip:bashton@sipgw.magrathea.net REGED internal profile sip:mod_sofia@192.168.196.101:5080 RUNNING (0) ================================================================================================= 2 profiles 0 aliases The SIP client is definitely listening: gdh@gdh-e7440:~$ netstat -anp | grep 5060 tcp 0 0 192.168.41.198:5060 0.0.0.0:* LISTEN 17566/qutecom udp 0 0 192.168.41.198:5060 0.0.0.0:* 17566/qutecom And 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/2.0 Via: SIP/2.0/UDP 192.168.196.101:5080;rport;branch=z9hG4bKX71j6K414rQNa Max-Forwards: 67 From: "None " <sip:07951357XXX@freeswitch.bashton.eu>;tag=rBeB1Nv1Bampm To: <sip:106@192.168.41.198:5060> Call-ID: f850b9a6-cb05-1232-7693-02b891c191f6 CSeq: 66131306 INVITE Contact: <sip:mod_sofia@192.168.196.101:5080> User-Agent: FreeSWITCH-mod_sofia/1.4.9~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 253 X-CALLINFO: cdr=5437AC53=94F6B14; X-FS-Support: update_display,send_info Remote-Party-ID: "None " <sip:07951357XXX@freeswitch.bashton.eu
;party=calling;screen=yes;privacy=off
The 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.