Directmedia/Canreinvite Not Working

Good morning Asterisk support community, I am really hoping you can help me out with this one. I am running FreePBX release 6.12.65-25 on Asterisk 11.16.0, and I cannot seem to get Directmedia to work. Basically, I would like the Asterisk to handle SIP messaging, but RTP to be passed directly from my host phones to my provider’s gateway across my SIP trunk. I am able to get this working with directrtpsetup=yes, however that doesn’t initiate a re-INVITE message with the new RTP source, and my carrier is doing media anchoring, so they toss RTP that comes from any source not defined in the SDP. Therefore, I must have directmedia=yes working, as it should re-INVITE with the phone’s IP as the media source, but that never happens. Although my hosts are on a different network than my Asterisk PBX, there is no NAT, everything is routed.

Here are my current general SIP settings:

[root@localhost asterisk]# vi sip_general_additional.conf
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-12.0.43(11.16.0)
disallow=all
allow=ulaw
tos=0x68
directmedia=nonat
keepalive=yes
canreinvite=yes
callevents=no
rtpend=20000
rtpstart=10000
jbenable=no
defaultexpiry=120
minexpiry=60
registerattempts=0
registertimeout=20
notifyhold=yes
notifyringing=yes
srvlookup=no
allowguest=yes
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=nonat
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
maxexpiry=3600
nat=no
ALLOW_SIP_ANON=no
externip=172.32.1.119
localnet=192.168.218.96/27
localnet=66.190.188.160/27
localnet=68.114.34.0/24
localnet=192.168.33.0/24
localnet=66.190.188.0/24
localnet=172.32.1.0/24
localnet=62.0.0.0/24
localnet=192.168.0.0/16

And here are my specific settings. I am testing using the 720-634-2073 host:

[3038356033]
deny=0.0.0.0/0.0.0.0
secret=3038356033aa
dtmfmode=rfc2833
canreinvite=yes
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/3038356033
mailbox=3038356033@default
permit=0.0.0.0/0.0.0.0
callerid=3038356033 <3038356033>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[7206342073]
secret=7206342073aa
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=yes
directmedia=yes
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=pai
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/7206342073
mailbox=7206342073@device
permit=0.0.0.0/0.0.0.0
callerid=7206342073 <7206342073>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[7206342074]
deny=0.0.0.0/0.0.0.0
secret=7206342074aa
dtmfmode=rfc2833
canreinvite=yes
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/7206342074
mailbox=7206342074@default
permit=0.0.0.0/0.0.0.0
callerid=7206342074 <7206342074>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[7206342115]
deny=0.0.0.0/0.0.0.0
secret=7206342115aa
dtmfmode=rfc2833
canreinvite=yes
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/7206342115
mailbox=7206342115@default
permit=0.0.0.0/0.0.0.0
callerid=7206342115 <7206342115>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

[sip_908_Multisubnet]
disallow=all
host=62.0.0.1
defaultuser=3038356033
secret=3038356033aa
type=peer
allow=ulaw
context=from-sip-external
insecure=port,invite
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
directmedia=nonat
canreinvite=yes
directmediapermit=0.0.0.0/0
keepalive=no

Here are the SIP debugs from a test outbound call. I know it isn’t working, as there is never a re-invite message changing the media IP to the IP of my phone (68.114.34.2). Also, RTP debugging showed the Asterisk was still handling it.

<— SIP read from UDP:68.114.34.2:5060 —>
INVITE sip:913038356007@172.32.1.119 SIP/2.0
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-43ad9433
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119
Remote-Party-ID: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;screen=yes;party=calling
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “Bob Mutisubnet Test” sip:7206342073@68.114.34.2:5060;ref=7206342073
Expires: 240
User-Agent: Cisco/SPA122-1.3.0(017_dbg_cdp3WmemIvrFix)
Content-Length: 246
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp

v=0
o=- 52860139 52860139 IN IP4 68.114.34.2
s=-
c=IN IP4 68.114.34.2
t=0 0
m=audio 16438 RTP/AVP 0 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 /8000
a=fmtp:0 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (15 headers 13 lines) —
Sending to 68.114.34.2:5060 (no NAT)
Sending to 68.114.34.2:5060 (no NAT)
Using INVITE request as basis request - 3322506-801d761c@68.114.34.2
Found peer ‘7206342073’ for ‘7206342073’ from 68.114.34.2:5060

<— Reliably Transmitting (no NAT) to 68.114.34.2:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-43ad9433;received=68.114.34.2
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as11d77ae4
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 101 INVITE
Server: FPBX-12.0.43(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="700fc583"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3322506-801d761c@68.114.34.2’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:68.114.34.2:5060 —>
ACK sip:913038356007@172.32.1.119 SIP/2.0
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-43ad9433
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as11d77ae4
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 101 ACK
Max-Forwards: 70
Contact: “Bob Mutisubnet Test” sip:7206342073@68.114.34.2:5060;ref=7206342073
User-Agent: Cisco/SPA122-1.3.0(017_dbg_cdp3WmemIvrFix)
Content-Length: 0

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

<— SIP read from UDP:68.114.34.2:5060 —>
INVITE sip:913038356007@172.32.1.119 SIP/2.0
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-e950a42f
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119
Remote-Party-ID: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;screen=yes;party=calling
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“7206342073”,realm=“asterisk”,nonce=“700fc583”,uri="sip:913038356007@172.32.1.119",algorithm=MD5,response="82c896b9ac8222362b108095d95dda42"
Contact: “Bob Mutisubnet Test” sip:7206342073@68.114.34.2:5060;ref=7206342073
Expires: 240
User-Agent: Cisco/SPA122-1.3.0(017_dbg_cdp3WmemIvrFix)
Content-Length: 246
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp

v=0
o=- 52860139 52860139 IN IP4 68.114.34.2
s=-
c=IN IP4 68.114.34.2
t=0 0
m=audio 16438 RTP/AVP 0 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 /8000
a=fmtp:0 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (16 headers 13 lines) —
Sending to 68.114.34.2:5060 (no NAT)
Using INVITE request as basis request - 3322506-801d761c@68.114.34.2
Found peer ‘7206342073’ for ‘7206342073’ from 68.114.34.2:5060
Found RTP audio format 0
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 - (ulaw), 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 68.114.34.2:16438
Looking for 913038356007 in from-internal (domain 172.32.1.119)
list_route: hop: sip:7206342073@68.114.34.2:5060;ref=7206342073

<— Transmitting (no NAT) to 68.114.34.2:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-e950a42f;received=68.114.34.2
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 INVITE
Server: FPBX-12.0.43(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:913038356007@172.32.1.119:5060
Content-Length: 0

<------------>
Audio is at 19924
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.0.0.1:5060:
INVITE sip:13038356007@62.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK20d7218c
Max-Forwards: 70
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1
Contact: sip:3038356033@172.32.1.119:5060
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 102 INVITE
User-Agent: FPBX-12.0.43(11.16.0)
Date: Mon, 16 Mar 2015 18:38:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: “3038356033” sip:3038356033@172.32.1.119;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 181595309 181595309 IN IP4 172.32.1.119
s=Asterisk PBX 11.16.0
c=IN IP4 172.32.1.119
t=0 0
m=audio 19924 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:62.0.0.1:5060 —>
SIP/2.0 100 Trying
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK20d7218c
Contact: sip:13038356007@62.0.0.1:5060;transport=UDP
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R10.3.0.E
Content-Length: 0

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

<— SIP read from UDP:62.0.0.1:5060 —>
SIP/2.0 180 Ringing
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1;tag=4407338-3e000001-13c4-fcac7-9831f676-fcac7
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK20d7218c
Contact: sip:13038356007@62.0.0.1:5060;transport=UDP
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R10.3.0.E
Content-Length: 0

<------------->
— (11 headers 0 lines) —
list_route: hop: sip:13038356007@62.0.0.1:5060;transport=UDP

<— Transmitting (no NAT) to 68.114.34.2:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-e950a42f;received=68.114.34.2
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 INVITE
Server: FPBX-12.0.43(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:913038356007@172.32.1.119:5060
Content-Length: 0

<— SIP read from UDP:62.0.0.1:5060 —>
SIP/2.0 200 OK
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1;tag=4407338-3e000001-13c4-fcac7-9831f676-fcac7
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK20d7218c
Contact: sip:13038356007@62.0.0.1:5060;transport=UDP
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R10.3.0.E
Content-Type: application/sdp
Content-Length: 177

v=0
o=BroadWorks 652 1 IN IP4 62.0.0.1
s=-
c=IN IP4 62.0.0.1
t=0 0
m=audio 10314 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
— (12 headers 10 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), 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 62.0.0.1:10314
list_route: hop: sip:13038356007@62.0.0.1:5060;transport=UDP
set_destination: Parsing sip:13038356007@62.0.0.1:5060;transport=UDP for address/port to send to
set_destination: set destination to 62.0.0.1:5060
Transmitting (no NAT) to 62.0.0.1:5060:
ACK sip:13038356007@62.0.0.1:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK6e3b0cee
Max-Forwards: 70
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1;tag=4407338-3e000001-13c4-fcac7-9831f676-fcac7
Contact: sip:3038356033@172.32.1.119:5060
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 102 ACK
User-Agent: FPBX-12.0.43(11.16.0)
Content-Length: 0


Audio is at 10756
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 68.114.34.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-e950a42f;received=68.114.34.2
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 INVITE
Server: FPBX-12.0.43(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:913038356007@172.32.1.119:5060
P-Asserted-Identity: “CID:3038356033” sip:13038356007@172.32.1.119
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1155156849 1155156849 IN IP4 172.32.1.119
s=Asterisk PBX 11.16.0
c=IN IP4 172.32.1.119
t=0 0
m=audio 10756 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

<------------>
Retransmitting #1 (no NAT) to 68.114.34.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-e950a42f;received=68.114.34.2
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 INVITE
Server: FPBX-12.0.43(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:913038356007@172.32.1.119:5060
P-Asserted-Identity: “CID:3038356033” sip:13038356007@172.32.1.119
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1155156849 1155156849 IN IP4 172.32.1.119
s=Asterisk PBX 11.16.0
c=IN IP4 172.32.1.119
t=0 0
m=audio 10756 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:68.114.34.2:5060 —>
ACK sip:913038356007@172.32.1.119:5060 SIP/2.0
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-d7745a2a
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“7206342073”,realm=“asterisk”,nonce=“700fc583”,uri="sip:913038356007@172.32.1.119",algorithm=MD5,response="82c896b9ac8222362b108095d95dda42"
Contact: “Bob Mutisubnet Test” sip:7206342073@68.114.34.2:5060;ref=7206342073
User-Agent: Cisco/SPA122-1.3.0(017_dbg_cdp3WmemIvrFix)
Content-Length: 0

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

<— SIP read from UDP:68.114.34.2:5060 —>
ACK sip:913038356007@172.32.1.119:5060 SIP/2.0
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-d7745a2a
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“7206342073”,realm=“asterisk”,nonce=“700fc583”,uri="sip:913038356007@172.32.1.119",algorithm=MD5,response="82c896b9ac8222362b108095d95dda42"
Contact: “Bob Mutisubnet Test” sip:7206342073@68.114.34.2:5060;ref=7206342073
User-Agent: Cisco/SPA122-1.3.0(017_dbg_cdp3WmemIvrFix)
Content-Length: 0

<— SIP read from UDP:68.114.34.2:5060 —>
BYE sip:913038356007@172.32.1.119:5060 SIP/2.0
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-f35a25c8
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username=“7206342073”,realm=“asterisk”,nonce=“700fc583”,uri=“sip:913038356007@172.32.1.119:5060”,algorithm=MD5,response="38fca0eeb706e1affb5aba0b02e4a346"
User-Agent: Cisco/SPA122-1.3.0(017_dbg_cdp3WmemIvrFix)
P-RTP-Stat: PS=167,OS=26720,PR=164,OR=26240,PL=0,JI=0,LA=0,DU=3,EN=G711u,DE=G711u
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 68.114.34.2:5060 (no NAT)
Got RTP packet from 62.0.0.1:10314 (type 00, seq 058564, ts 3940653611, len 000160)
Sent RTP packet to 68.114.34.2:16438 (type 00, seq 048398, ts 3940653608, len 000160)
Scheduling destruction of SIP dialog ‘3322506-801d761c@68.114.34.2’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 68.114.34.2:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 68.114.34.2:5060;branch=z9hG4bK-f35a25c8;received=68.114.34.2
From: “Bob Mutisubnet Test” sip:7206342073@172.32.1.119;tag=4c0167fe81024f84o0
To: sip:913038356007@172.32.1.119;tag=as67697045
Call-ID: 3322506-801d761c@68.114.34.2
CSeq: 103 BYE
Server: FPBX-12.0.43(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:13038356007@62.0.0.1:5060;transport=UDP for address/port to send to
set_destination: set destination to 62.0.0.1:5060
Reliably Transmitting (no NAT) to 62.0.0.1:5060:
BYE sip:13038356007@62.0.0.1:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK25ce1440
Max-Forwards: 70
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1;tag=4407338-3e000001-13c4-fcac7-9831f676-fcac7
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 103 BYE
User-Agent: FPBX-12.0.43(11.16.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:62.0.0.1:5060 —>
SIP/2.0 200 OK
From: sip:3038356033@172.32.1.119;tag=as56ac604b
To: sip:13038356007@62.0.0.1;tag=4407338-3e000001-13c4-fcac7-9831f676-fcac7
Call-ID: 71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060
CSeq: 103 BYE
Via: SIP/2.0/UDP 172.32.1.119:5060;branch=z9hG4bK25ce1440
Supported: 100rel,replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R10.3.0.E
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘71d1928e1bd4865628e51b2d145098c6@172.32.1.119:5060’ Method: INVITE
localhost*CLI>

You have conflicting settings for the deprecated and and current synonyms of the same setting.

Direct media can be inhibited by lots of things in the dialplan. As FreePBX controls that, you need to ask on community.freepbx.org/