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 [WARNING] mod_dptools.c:1628 incoming call from [07951357XXX] to [106]
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:
=================================================================================================
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
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
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
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
=================================================================================================
=================================================================================================
2 profiles 0 aliases
The SIP client is definitely listening:
gdh@gdh-e7440:~$ netstat -anp | grep 5060
And I can see that when the call comes in, there is traffic received by the SIP client (tcpdump + wireshark):
Via: SIP/2.0/UDP 192.168.196.101:5080;rport;branch=z9hG4bKX71j6K414rQNa
Max-Forwards: 67
From: "None
Call-ID: f850b9a6-cb05-1232-7693-02b891c191f6
CSeq: 66131306 INVITE
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
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.