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>