Inbound Calls Drop After 10 Seconds: Retransmission Error

@luongjames wrote:

Hello - I'm new to FreePBX. I've searched the forums a bit here but haven't found a good solution.

Here's the situation:
- Outbound Calls are Fine
- Inbound Calls drop after about 10 seconds
- Port Forwarding of 5060 and 10001-20000 enabled to FreePBX server
- NAT Enabled

The error that I get is:

Really destroying SIP dialog '15fcac6a097f067b064316761a41115a@10.0.3.10:5160' Method: INVITE
[2017-02-06 15:39:55] WARNING[2187]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 2009444203@192.168.1.211 for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2017-02-06 15:39:55] WARNING[2187]: chan_sip.c:4085 retrans_pkt: Hanging up call 2009444203@192.168.1.211 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Really destroying SIP dialog '7fb49746397917454b14d96059b8cffd@10.0.3.10:5160' Method: INVITE
Scheduling destruction of SIP dialog '14c3e7076d4ff7fb3a5017337fc3f8b0@10.0.3.10:5160' in 6400 ms (Method: INVITE)
  == Spawn extension (macro-dial, s, 19) exited non-zero on 'SIP/31507580-0000008d' in macro 'dial'
set_destination: Parsing <sip:801@10.0.3.7> for address/port to send to
set_destination: set destination to 10.0.3.7:5060
Reliably Transmitting (no NAT) to 10.0.3.7:5060:
BYE sip:801@10.0.3.7 SIP/2.0
Via: SIP/2.0/UDP 10.0.3.10:5160;branch=z9hG4bK24a70dce
Max-Forwards: 70
From: "83860122" <sip:83860122@10.0.3.10:5160>;tag=as4a49841f
To: <sip:801@10.0.3.7>;tag=28E31BD7-687C9964
Call-ID: 14c3e7076d4ff7fb3a5017337fc3f8b0@10.0.3.10:5160
CSeq: 103 BYE
User-Agent: FPBX-13.0.190.11(13.12.1)
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

I looked at the SIP DEBUG to look further at whats happening and this is what I get.

---
Retransmitting #3 (NAT) to 203.117.31.248:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.117.31.248;branch=z9hG4bK0152.156dbed7.0;received=203.117.31.248;rport=5060
Via: SIP/2.0/UDP 192.168.1.211:6889;rport=6889;branch=z9hG4bK21822095
Record-Route: <sip:203.117.31.248;r2=on;ftag=239907731;lr=on>
Record-Route: <sip:192.168.1.179;r2=on;ftag=239907731;lr=on>
From: 83860122 <sip:83860122@sip.pfingo.com:5060>;tag=239907731
To: <sip:xxxxxxxxxx@sip.pfingo.com:5060>;tag=as6f96bf07
Call-ID: 2009444203@192.168.1.211
CSeq: 20 INVITE
Server: FPBX-13.0.190.11(13.12.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:600@175.156.75.129:5160>
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 1450624280 1450624280 IN IP4 175.156.75.129
s=Asterisk PBX 13.12.1
c=IN IP4 175.156.75.129
t=0 0
m=audio 16748 RTP/AVP 0 110
a=rtpmap:0 PCMU/8000
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

Would anyone have suggestions on how I can resolve this? I did notice that there is a private IP string within the error message 192.168.1.xxx. I am not using this subnet on any of my internal networks. Not sure how this got there, and whether this is affecting the problem at all.

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