INVITE timeout on incoming calls

I am using FreePBX, with Asterisk 10.2.0. I have a SIP trunk that all calls come in from and go out of. All outgoing calls work without any issue, but with the incoming calls my PBX is not accepting the ACK sent from the SIP trunk provider. My PBX just keeps sending 200 OK messages and in the asterisk console I see the ACK responses but after that the PBX resends 200 OK like it never got the ACK. Below is the debug information that I am getting

[code] == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 19736
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to internalSIP:5060:
INVITE sip:201@internalSIP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK29073a10;rport
Max-Forwards: 70
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp
Contact: sip:numberCalledFrom@PBXaddress:5060
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(10.2.0)
Date: Mon, 23 Apr 2012 20:48:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 145269813 145269813 IN IP4 PBXaddress
s=Asterisk PBX 10.2.0
c=IN IP4 PBXaddress
t=0 0
m=audio 19736 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/201

<— Transmitting (NAT) to 67.216.45.45:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Length: 0

<------------>
Retransmitting #1 (NAT) to internalSIP:5060:
INVITE sip:201@internalSIP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK29073a10;rport
Max-Forwards: 70
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp
Contact: sip:numberCalledFrom@PBXaddress:5060
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.10.0(10.2.0)
Date: Mon, 23 Apr 2012 20:48:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 145269813 145269813 IN IP4 PBXaddress
s=Asterisk PBX 10.2.0
c=IN IP4 PBXaddress
t=0 0
m=audio 19736 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:internalSIP:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK29073a10;rport
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:201@internalSIP:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:internalSIP:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK29073a10;rport
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:201@internalSIP:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:internalSIP:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK29073a10;rport
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp;tag=00055e7cd6d0564f659f1f20-630a845e
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:201@internalSIP:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “extnesion201” sip:201@PBXaddress;party=called;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

<------------->
— (11 headers 0 lines) —
list_route: hop: sip:201@internalSIP:5060;transport=udp
– SIP/201-00000050 is ringing

<— Transmitting (NAT) to 67.216.45.45:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Length: 0

<------------>

<— SIP read from UDP:internalSIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK29073a10;rport
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp;tag=00055e7cd6d0564f659f1f20-630a845e
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 INVITE
Server: Cisco-CP7960G/8.0
Contact: sip:201@internalSIP:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “brett” sip:201@sip.get2spec.com;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 199
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 20698 0 IN IP4 internalSIP
s=SIP Call
t=0 0
m=audio 18562 RTP/AVP 0 101
c=IN IP4 internalSIP
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port internalSIP:18562
list_route: hop: sip:201@internalSIP:5060;transport=udp
set_destination: Parsing sip:201@internalSIP:5060;transport=udp for address/port to send to
set_destination: set destination to internalSIP:5060
Transmitting (NAT) to internalSIP:5060:
ACK sip:201@internalSIP:5060;transport=udp SIP/2.0

Transmitting (NAT) to internalSIP:5060:
ACK sip:201@internalSIP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK190503e6;rport
Max-Forwards: 70
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp;tag=00055e7cd6d0564f659f1f20-630a845e
Contact: sip:numberCalledFrom@PBXaddress:5060
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.0(10.2.0)
Content-Length: 0


-- SIP/201-00000050 answered SIP/gafachi1b-0000004f

Audio is at 18294
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 67.216.45.45:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bK5c765def
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Retransmitting #1 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKf0c3f901
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Retransmitting #2 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bK9f3f8cbf
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->

Retransmitting #3 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKf791af92
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->

Really destroying SIP dialog ‘6cd1f6ae42dece13234c817b6f44d6ec@PBXaddress:5060’ Method: OPTIONS
Retransmitting #4 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
— (10 headers 0 lines) —
Retransmitting #5 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bK9f0dba96
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Retransmitting #6 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKcb1219a9
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0


Retransmitting #7 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bK78dbd953
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0


Really destroying SIP dialog ‘63d6d7c936c074c0416c93f85318de6a@PBXaddress:5060’ Method: OPTIONS
Retransmitting #8 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKd5b8937a
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Retransmitting #9 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bK5822fc92
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0


Retransmitting #10 (NAT) to 67.216.45.45:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bKce773f2c;received=67.216.45.45;rport=5060
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as3564545c
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 INVITE
Server: FPBX-2.10.0(10.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:DIDnumber@externalIP:5060
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 803221021 803221021 IN IP4 externalIP
s=Asterisk PBX 10.2.0
c=IN IP4 externalIP
t=0 0
m=audio 18294 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:67.216.45.45:5060 —>
ACK sip:DIDnumber@externalIP:5060 SIP/2.0
Via: SIP/2.0/UDP 67.216.45.45:5060;branch=z9hG4bK9e025d09
From: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
To: sip:DIDnumber@externalIP;tag=as03cac74c
Contact: sip:numberCalledFrom@67.216.45.45
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 103 ACK
User-Agent: Gafachi UAC v110.09
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2012-04-23 13:49:31] WARNING[-1]: chan_sip.c:3663 retrans_pkt: Retransmission timeout reached on transmission e170929be9aee73869bf1b0081ed3ed2@67.216.45.45 for seqno 103 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[2012-04-23 13:49:31] WARNING[-1]: chan_sip.c:3692 retrans_pkt: Hanging up call e170929be9aee73869bf1b0081ed3ed2@67.216.45.45 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
– Executing [h@macro-dial-one:1] Macro(“SIP/gafachi1b-0000004f”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/gafachi1b-0000004f”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] Hangup(“SIP/gafachi1b-0000004f”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/gafachi1b-0000004f’ in macro ‘hangupcall’
== Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/gafachi1b-0000004f’
Scheduling destruction of SIP dialog ‘55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:201@internalSIP:5060;transport=udp for address/port to send to
set_destination: set destination to internalSIP:5060
Reliably Transmitting (NAT) to internalSIP:5060:
BYE sip:201@internalSIP:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP PBXaddress:5060;branch=z9hG4bK2d150073;rport
Max-Forwards: 70
From: “unknown” sip:numberCalledFrom@PBXaddress;tag=as305a1fd3
To: sip:201@internalSIP:5060;transport=udp;tag=00055e7cd6d0564f659f1f20-630a845e
Call-ID: 55ee2b703c7d7a52153ce9a36892e862@PBXaddress:5060
CSeq: 103 BYE
User-Agent: FPBX-2.10.0(10.2.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


== Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/gafachi1b-0000004f’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘SIP/gafachi1b-0000004f’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 201, 2) exited non-zero on 'SIP/gafachi1b-0000004f’
Scheduling destruction of SIP dialog ‘e170929be9aee73869bf1b0081ed3ed2@67.216.45.45’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:numberCalledFrom@67.216.45.45 for address/port to send to
set_destination: set destination to 67.216.45.45:5060
Reliably Transmitting (NAT) to 67.216.45.45:5060:
BYE sip:numberCalledFrom@67.216.45.45 SIP/2.0
Via: SIP/2.0/UDP externalIP:5060;branch=z9hG4bK365d67b2;rport
Max-Forwards: 70
From: sip:DIDnumber@externalIP;tag=as3564545c
To: “unknown” sip:numberCalledFrom@sip1b.gafachi.com;tag=gss6872a3d8235a6a91aeb4175c67742768
Call-ID: e170929be9aee73869bf1b0081ed3ed2@67.216.45.45
CSeq: 102 BYE
User-Agent: FPBX-2.10.0(10.2.0)
Proxy-Authorization: Digest username=“a9971sRU5po76YTJ”, realm=“asterisk”, algorithm=MD5, uri=“sip:externalIP”, nonce="", response="15f270aa14139263924e6b6a0468f006"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


[/code]

I did read in the mailing list that there was someone who saw the same issue but I did not see any response ever posted. I have not found anything else related to this issue other than that post on the Asterisk-Users mailing list which indicated that it was something specific to Asterisk 1.8. I have tried unsuccessfully to get anything older than Asterisk 1.8 to work with my trunk provider, and the provider will not give me any indication on what is going on, or try to help other than to say that it is something on my end.

Does anyone have any idea of what could be causing this problem, or how I can fix it?

The To tag doesn’t match. The peer is broken. You might want to see if disabling strict SIP processing helps.

[quote=“RFC3261”]15.1.1 UAC Behavior

A BYE request is constructed as would any other request within a
dialog, as described in Section 12.
[/quote]

[quote=“RFC3261”]12.2.1.1 Generating the Request

A request within a dialog is constructed by using many of the
components of the state stored as part of the dialog.

The URI in the To field of the request MUST be set to the remote URI
from the dialog state. The tag in the To header field of the request
[color=#FF0000]MUST[/color] be set to the remote tag of the dialog ID. The From URI of the[/quote]

Thanks, that seems to have fixed it. Do you know why this would be required? It seems like Gafachi who is providing the trunk is doing some very strange things on their end requiring me to create workarounds on my end to get it to work. An example that is not shown here is I had to set the fromdomain to be something different from the host address, they told me that it does not matter what it is set to as long as it is not the same as the host address, and the fromdomain seems to default to being the same as the host address.

Also is it possible that a strange configuration on their end is causing things like Trixbox which is using Asterisk 1.6 to not work with Gafachi even when it is setup exactly the same as a working version of freePBX?

All I know is that I have had all kinds of problems getting this Gafachi trunk working and at home I recently got service from Flowroute and got it working in less than 5 minutes without needing any kind of support.

Hi,

I am facing the same issue. Could you please let me know where, in which file you disabled strict sip processing.

Thanks in advance
Bipin

I am having the exact same issue with Gafachi? Anyone can help me on how to disable the strict sip processing? Any help is greatly appreciated in advance. Thanks.

[quote];pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to “yes”)[/quote]

As to the file, that is so obvious that I’m afraid I have to leave you to work it out for yourself. If you don’t have the sample configuration files, you should download the sources and take a copy of them, as they are a key part of the documentation.

Any conflict with a GUI should be addressed on the forums for that GUI.

Thank you, just did not know it was the pedantic option. I know where it is and already has been changed. Hope it fixes the issue with Gafachi. Thanks again.