Incoming calls drop @ 30 seconds on FPBX-2.9.0(1.8.7.0)

Greetings to all!

I have finally have a 99% working server but incoming calls are being dropped. I have Googled and found no solutions that really address this issue. Firewall has all the required settings based on everything I have read. Can someone be so kind as to shed some light on the subject here. Please see bellow and thank you in advance.

[2011-12-01 14:18:00] VERBOSE[29418] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[2011-12-01 14:18:00] VERBOSE[29418] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2011-12-01 14:18:00] VERBOSE[29418] chan_sip.c:
<— Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[2011-12-01 14:18:00] VERBOSE[2792] chan_sip.c: Retransmitting #1 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:01] VERBOSE[2792] chan_sip.c: Retransmitting #2 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:03] VERBOSE[2792] chan_sip.c: Retransmitting #3 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:05] VERBOSE[29420] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2011-12-01 14:18:05] VERBOSE[29420] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c: Reliably Transmitting (no NAT) to xxx.xx.xxx.xxx:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK48f4a7cb
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as73bfc3bd
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.7.0)
Date: Thu, 01 Dec 2011 19:18:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xxx.xx.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK48f4a7cb
From: “Unknown” sip:[email protected];tag=as73bfc3bd
To: “Ramon F McDougall” sip:[email protected]:5060;tag=F72BC885-E820DD40
CSeq: 102 OPTIONS
Call-ID: [email protected]:5060
Contact: sip:[email protected]:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Supported: 100rel,replaces,100rel,timer,replaces,norefersub
User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.4.0267
Accept-Language: en
Accept: application/sdp,text/plain,message/sipfrag,application/dialog-info+xml
Accept-Encoding: identity
Content-Length: 0

<------------->
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c: — (14 headers 0 lines) —
[2011-12-01 14:18:05] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2011-12-01 14:18:06] VERBOSE[29422] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2011-12-01 14:18:07] VERBOSE[29422] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[2011-12-01 14:18:07] VERBOSE[2792] chan_sip.c: Retransmitting #4 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:11] VERBOSE[2792] chan_sip.c: Retransmitting #5 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:15] VERBOSE[2792] chan_sip.c: Retransmitting #6 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:19] VERBOSE[2792] chan_sip.c: Retransmitting #7 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: NWHN6TH4HRGCTPMGA36[email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:20] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: Retransmitting #8 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xx.xx.xx.xx:5060 —>
OPTIONS sip:xx.xxx.xx.xx:1345 SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=0
From: sip:[email protected];tag=fc6826d1
To: sip:xx.xxx.xx.xx:1345
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: — (7 headers 0 lines) —
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: Looking for s in from-sip-external (domain xx.xxx.xx.xx:1345)
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c:
<— Transmitting (NAT) to xx.xx.xx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=0;received=xx.xx.xx.xx;rport=5060
From: sip:[email protected];tag=xxxxxxxx
To: sip:xx.xxx.xx.xx:1345;tag=as2342c90b
Call-ID: [email protected]
CSeq: 1 OPTIONS
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:xx.xxx.xx.xx:5060
Accept: application/sdp
Content-Length: 0

<------------>
[2011-12-01 14:18:23] VERBOSE[2792] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)
[2011-12-01 14:18:27] VERBOSE[2792] chan_sip.c: Retransmitting #9 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:31] VERBOSE[29427] manager.c: == Manager ‘admin’ logged on from 127.0.0.1
[2011-12-01 14:18:31] VERBOSE[29427] manager.c: == Manager ‘admin’ logged off from 127.0.0.1
[2011-12-01 14:18:31] VERBOSE[2792] chan_sip.c: Retransmitting #10 (no NAT) to xx.xx.xx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKc8ee.6a141044.0;received=xx.xx.xx.xx
Via: SIP/2.0/UDP 81.201.84.195:5060;branch=z9hG4bKd9e7b4a4e8e8edeb6ea2e7d9f8751579
Record-Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
From: “18005551212” sip:[email protected];tag=68693
To: sip:[email protected];tag=as2b11125f
Call-ID: [email protected]
CSeq: 102 INVITE
Server: FPBX-2.9.0(1.8.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 2053286403 2053286403 IN IP4 xx.xxx.xx.xx
s=Asterisk PBX 1.8.7.0
c=IN IP4 xx.xxx.xx.xx
t=0 0
m=audio 16954 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[2011-12-01 14:18:32] WARNING[2792] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2011-12-01 14:18:32] WARNING[2792] chan_sip.c: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Executing [[email protected]:1] Macro(“SIP/VoipVoip-00000075”, “hangupcall,”) in new stack
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/VoipVoip-00000075”, “1?theend”) in new stack
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Goto (macro-hangupcall,s,3)
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: – Executing [[email protected]:3] Hangup(“SIP/VoipVoip-00000075”, “”) in new stack
[2011-12-01 14:18:32] VERBOSE[29418] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/VoipVoip-00000075’ in macro ‘hangupcall’
[2011-12-01 14:18:32] VERBOSE[29418] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/VoipVoip-00000075’
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE)
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: Parsing sip:[email protected]:5060 for address/port to send to
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: set destination to xxx.xx.xxx.xxx:5060
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Reliably Transmitting (no NAT) to xxx.xx.xxx.xxx:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK1e85e8ef
Max-Forwards: 70
From: “18005551212” sip:[email protected];tag=as1a25bd1e
To: sip:[email protected]:5060;tag=F502CC4-D8002607
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: FPBX-2.9.0(1.8.7.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


[2011-12-01 14:18:32] VERBOSE[29418] app_macro.c: == Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/VoipVoip-00000075’ in macro ‘dial-one’
[2011-12-01 14:18:32] VERBOSE[29418] app_macro.c: == Spawn extension (macro-exten-vm, s, 7) exited non-zero on ‘SIP/VoipVoip-00000075’ in macro ‘exten-vm’
[2011-12-01 14:18:32] VERBOSE[29418] pbx.c: == Spawn extension (from-did-direct, 2000, 2) exited non-zero on ‘SIP/VoipVoip-00000075’
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: Parsing sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on for address/port to send to
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: set_destination: set destination to xx.xx.xx.xx:5060
[2011-12-01 14:18:32] VERBOSE[29418] chan_sip.c: Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xx.xx:5060;branch=z9hG4bK1a603d48
Route: sip:xx.xx.xx.xx;ftag=68693;nat=yes;lr=on
Max-Forwards: 70
From: sip:[email protected];tag=as2b11125f
To: “18005551212” sip:[email protected];tag=68693
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-2.9.0(1.8.7.0)
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xxx.xx.xxx.xxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP FreePBXIPADD:5060;branch=z9hG4bK1e85e8ef
From: “18005551212” sip:[email protected];tag=as1a25bd1e
To: “Ramon F McDougall” sip:[email protected]:5060;tag=F502CC4-D8002607
CSeq: 103 BYE
Call-ID: [email protected]:5060
Contact: sip:[email protected]:5060
User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.4.0267
Accept-Language: en
Content-Length: 0

<------------->
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: — (10 headers 0 lines) —
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c:
<— SIP read from UDP:xx.xx.xx.xx:5060 —>
SIP/2.0 200 OK
Record-Route: sip:xx.xx.xx.xx;ftag=as2b11125f;lr=on
Via: SIP/2.0/UDP xx.xxx.xx.xx:5060;branch=z9hG4bK1a603d48;rport=1345
From: sip:[email protected];tag=as2b11125f
To: “18005551212” sip:[email protected];tag=68693
Call-ID: [email protected]
CSeq: 102 BYE
Content-Length: 0

<------------->
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: — (8 headers 0 lines) —
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[2011-12-01 14:18:32] VERBOSE[2792] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: INVITE
[2011-12-01 14:18:32] VERBOSE[29429] manager.c: == Manager ‘admin’ logged on from 127.0.0.1

Thank you all!

If you google this site there are over 100 threads on short call dropped audio.

It’s a NAT/RTP issue, it’s always a NAT issue.

I don’t want to sound harsh but everyone seems to suffer from terminal uniqueness when the it’s always the same root cause.

If it’s a point to point connection make sure that Asterisk is not translating (appropriate localnet setttings). If it is NAT’ing make sure the externIP is setup correctly.

That’s all it takes.

Try to set “Reinvite Behavior” to yes in your SIP configuration.

I had a similar problem with Euteliavoip.it and solved it enabling Reivite.

I will recheck NAT settings but my settings agree with those 100 posts. Re-invite is set to “yes”.

Your comments are greatly appreciated.

Regards

You can try adding

qualifyfreq=25

as OTHER SIP settings

With the SIP setting module.

I tried those settings, makes no difference.

Thank you.

How can I undo qualifyfreq=25? Would this have an adverse effect on the system?

Thanks

I’ve noticed a severe tendency for people in the business of writing code or applications with their utter lack of ability to explain things in plain English. For those of you who live in the real world and would like this issue explained in plain English – here it goes.

First of all, SkykingOH was absolutely right in that is is a NAT issue. Ignore the rest of the punditry trying to lead you down the other paths like reinvites, etc. Its all crap. It really is just the NAT configuration but specifically here it is.

The freePBX box need to know whether you need to NAT from public IP addresses to public internal addresses since it doesn’t know if you’ve configured the ethernet inteface of the freePBX box on a public ip address directly (really bad idea BTW) or on a private internal address that is having a public IP address forwarded to it (like a VIP address or SNAT/DNAT firewall configuration). Ideally this second option is the preferred configuration.

Either way it needs to know what environment it is in on your network. If the ethernet interface is set on a private address that is accessible from the internet on a public address (via SNAT/DNAT, VIP, or whatever from a separate real router) then the NAT setting INSIDE the freePBX needs to think it is on a public IP address. This is also true if you really did set your ethernet interface on a real public IP address.

Quick note, at this point if you have set your interface on a real public ip address then now is a good time to stop doing that.

If you really only have the ethernet interface on a private IP address (like if the system doesn’t really go out to the internet) then freePBX needs to know that its NAT setting is NOT public and needs one of the alternative NAT configurations.

Here are the settings for the current 2.9.0.7 version of FreePBX.

In the GUI (Web Interface you log into to change things in FreePBX), go to:

  1. Tools, Advanced Administration, Asterisk SIP settings.

  2. For IP Configuration select public, static, or dynamic IP. (If you have the SNAT/DNAT, VIP, etc from your real router this is where you select public.)

  3. Make sure to save the config and click that little orange color tab that shows up at the top of the screen to reload the configuration. After that all should work well and you can go back to doing real things.

Pundity, I like that, too funny.

The funny thing is your message will be above someone who does not know vendor specific terms like Juniper’s MIP/VIP or what DNAT is. How far down do you have to go.

To me the Asterisk NAT exclusion rules are fundamentally no different than NAT exclusion rules you have to do on VPN routes to ignore interesting traffic. The concepts are all the same. It is also explained very will in the sample sip.conf distributed with Asterisk.

Further, if you don’t understand NAT, can’t read the Asterisk DOCS and translate them to Asterisk + FreePBX perhaps you should not be trying to integrate your own IP phones system or God forbid charging some unsuspecting schlep to do it for them. People wonder why I bristle when I see the word “client” or any other clue indicating you are doing this for hire then coming to us to help bail your ass out of a jam.

Just because the software is Free does not make it easy, everything comes at a price. If you pay Cisco prices for CUCM I expect them to have a team of tech writers writing highly detailed practices for every procedure. In the open source would the folks that make this possible have to write, test and maintain the code, document it, answer questions in the forum with some degree of civility (I often fall far short of this goal) and still have to make a living to support ourselves!

You know I feel better already for getting that off my chest. I am also very pleased that my little pointer got you on the right track and you are getting some use out of the system.

I really appreciate you taking the time to explain the context under which FreePBX works. The system is not dropping the calls at 30 secs.

I changed my settings, I kept NAT=no and set IP=Public. Initially I had the configuration where I specified the outside IP and the internal network with its netmask.That did not work even if I set NAT=yes.

Again,thanks for your contribution.

cyber - I thought that I was commenting on the OP.

So you are telling us in general terms but nothing specific.

Using the code tags to make your code readable (see input format below the text box). Post the output of ‘sip show settings’ also ‘sip show peer xxx’ where xxx is the name of your trunk.

Then out of sip_additional.conf I need the information from the trunk your are having issues with (again with the code tags).

Of course sanitize any private info, but no the first three octects of any IP’s

Hi! is this solved already?

Thanks squarejester for your helpful explanation. You helped me get a working setup.
I find the NAT settings…temperamental. I have had a working setup…then it stops working…then I fiddle with these settings, and it works again… I even found that the system worked better ( two way audio rather than one way) when I deliberately put the wrong external ip in!
Final solution ( seems to work for now - hoping it stays that way!) for me was to tell the system I had a public ip (although, actually, it is behind nat, so I had been using the setting ‘static’) and set NAT to ‘yes’.
Using ‘static’ and ‘fiddling with nat=no’ , ‘nat=yes’ etc, I had been able to get EITHER working outgoing calls ( two way audio, no 30 sec dropout) or working incoming calls, but the other (incoming, or outgoing respectively) would be one way audio, and drop after 30 secs. The breakthrough was switching to ‘public’ for the ip.
For those interested, i have my system behind nat, and NO port forwarding from outside in. Outside inw. calls are via the ‘phone no’ issued to me by my voip supplier. I do have a static ip for my ‘outside’ address, but this is on my router, and my pabx is behind NAT. (while I don’t have port forwarding, my router tracks and forwards returning traffic initiated from inside LAN).

i have the same issue, this happens with DAHDI or SIP, when my PRI provider sends the call over SIP (when dahdi breaks) it still goes through a ring group, after a while, if the call comes in dahdi, it’ll say call clearing cause 16, it will continue ringing from my cell phone, console will show the call reappearing, going through the process of the ring group again but this time completing the ring group, this happens about 20-25 seconds into the hunt.

any suggestions ?

nat is off. everything is on a LAN however i do have externip set up for failover outgoing SIP calling.

Hi Team,

Kindly suggest me as i am facing this issue from last 4 days.Whenever i am doing call from my Sip phone.Call is transferring to my defined context but after 30 seconds it is automatically disconnecting. Find below sip debugging logs for the same.

Retransmitting #8 (NAT) to 14.98.152.152:14790:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 14.98.152.152:14790;branch=z9hG4bK-d8754z-016449b0fa671442-1—d8754z-;received=14.98.152.152;rport=14790
From: "987654"sip:[email protected];tag=93659dd5
To: "900620001"sip:[email protected];tag=as6235b3a7
Call-ID: NGU3YWUxZmRiYzM0NGRlZjAwZTJkNWZlNDhlYjViNTQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1074757620 1074757620 IN IP4 10.10.2.10
s=Asterisk PBX 1.8.13.0
c=IN IP4 10.10.2.10
t=0 0
m=audio 11564 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


Retransmitting #9 (NAT) to 14.98.152.152:14790:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 14.98.152.152:14790;branch=z9hG4bK-d8754z-016449b0fa671442-1—d8754z-;received=14.98.152.152;rport=14790
From: "987654"sip:[email protected];tag=93659dd5
To: "900620001"sip:[email protected];tag=as6235b3a7
Call-ID: NGU3YWUxZmRiYzM0NGRlZjAwZTJkNWZlNDhlYjViNTQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1074757620 1074757620 IN IP4 10.10.2.10
s=Asterisk PBX 1.8.13.0
c=IN IP4 10.10.2.10
t=0 0
m=audio 11564 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #10 (NAT) to 14.98.152.152:14790:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 14.98.152.152:14790;branch=z9hG4bK-d8754z-016449b0fa671442-1—d8754z-;received=14.98.152.152;rport=14790
From: "987654"sip:[email protected];tag=93659dd5
To: "900620001"sip:[email protected];tag=as6235b3a7
Call-ID: NGU3YWUxZmRiYzM0NGRlZjAwZTJkNWZlNDhlYjViNTQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 1074757620 1074757620 IN IP4 10.10.2.10
s=Asterisk PBX 1.8.13.0
c=IN IP4 10.10.2.10
t=0 0
m=audio 11564 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

== Spawn extension (from-pstn, h, 10) exited non-zero on 'SIP/987654-00000001’
Scheduling destruction of SIP dialog ‘NGU3YWUxZmRiYzM0NGRlZjAwZTJkNWZlNDhlYjViNTQ.’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:14790 for address/port to send to
set_destination: set destination to 14.98.152.152:14790
Reliably Transmitting (NAT) to 14.98.152.152:14790:
BYE sip:[email protected]:14790 SIP/2.0
Via: SIP/2.0/UDP 10.10.2.10:5060;branch=z9hG4bK107c8e1d;rport
Max-Forwards: 70
From: “900620001"sip:[email protected];tag=as6235b3a7
To: “987654"sip:[email protected];tag=93659dd5
Call-ID: NGU3YWUxZmRiYzM0NGRlZjAwZTJkNWZlNDhlYjViNTQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.13.0
Proxy-Authorization: Digest username=“987654”, realm=“asterisk”, algorithm=MD5, uri=“sip:41.139.138.44”, nonce=””, response="158cc934df221cd60f883dc978573bc0"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0

<— SIP read from UDP:14.98.152.152:14790 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.2.10:5060;branch=z9hG4bK107c8e1d;rport=5060;received=41.139.138.44
Contact: sip:[email protected]:14790
To: "987654"sip:[email protected];tag=93659dd5
From: "900620001"sip:[email protected];tag=as6235b3a7
Call-ID: NGU3YWUxZmRiYzM0NGRlZjAwZTJkNWZlNDhlYjViNTQ.
CSeq: 102 BYE
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0