4G cellphone and asterisk behind nat problem

I have a problem of audio and outbound calling. let me describe the network I have:

4G cellphone<---------------->ADSR router<---------WiFi------>Asterisk<--------WiFi----->cellphone

sip.conf

;sip.conf
[general]
realm=kajs.homelinux.com; dyndns
udpbindaddr=0.0.0.0;
transport=udp,ws
bindport=5061
localnet=192.168.178.0/24
directmedia=no

[1064] ; This will be the legacy SIP client
type=friend
username=1064
host=dynamic
secret=1234
qualify=no
context=default

[1066] ; This will be the legacy SIP client
type=friend
username=1066
host=dynamic
secret=1234
context=default
directmedia=no
icesupport=yes
qualify=no
nat=force_rpot,comedia
disallow=all
allow=ulaw

sip debug on

To: “1066” sip:1066@kabejames.homelinux.com:5061
Call-ID: 93a6076779b78731
CSeq: 281493831 REGISTER
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“1066”,realm=“kabejames.homelinux.com”,nonce=“4ce1a1ff”,uri=“sip:kabejames.homelinux.com:5061”,response=“6ff303dbdb2d3407ffea45966e7c39eb”,algorithm=MD5
Contact: sip:1066@10.82.203.71:5060;expires=120
User-Agent: Media5-fone/3.7.1.1118
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 185.40.62.230:5060 (NAT)

<— Transmitting (no NAT) to 185.40.62.230:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.82.203.71;branch=z9hG4bKee63e800599edc3e8;received=185.40.62.230
From: “1066” sip:1066@kabejames.homelinux.com:5061;tag=c766b04da5
To: “1066” sip:1066@kabejames.homelinux.com:5061;tag=as667e99d7
Call-ID: 93a6076779b78731
CSeq: 281493831 REGISTER
Server: Asterisk PBX 12.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: sip:1066@10.82.203.71:5060;expires=120
Date: Mon, 15 Dec 2014 21:03:45 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘93a6076779b78731’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:94.252.40.87:5060 —>
INVITE sip:1066@kabejames.homelinux.com:5061 SIP/2.0
Accept: application/conference-info+xml, application/sdp, message/sipfrag, multipart/mixed
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bKcbe1c2fd400edb4d7
Max-Forwards: 70
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061
Call-ID: 9bff9ef080d431aa
CSeq: 860568219 INVITE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Contact: sip:1064@192.168.178.70:5060;audio
Supported: 100rel, replaces, sdp-anat, join
User-Agent: Media5-fone/4.1.2.2653 iOS/7.1.2
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 306

v=0
o=- 3251547188403486068 3251547188403486069 IN IP4 192.168.178.70
s=m5
c=IN IP4 192.168.178.70
t=0 0
a=sendrecv
m=audio 10000 RTP/AVP 0 8 96 125
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:96 mode=20
a=fmtp:125 0-15
a=sendrecv
<------------->
— (16 headers 14 lines) —
Sending to 94.252.40.87:5060 (NAT)
Sending to 94.252.40.87:5060 (NAT)
Using INVITE request as basis request - 9bff9ef080d431aa
Found peer ‘1064’ for ‘1064’ from 94.252.40.87:5060

<— Reliably Transmitting (NAT) to 94.252.40.87:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bKcbe1c2fd400edb4d7;received=94.252.40.87;rport=5060
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061;tag=as164560ca
Call-ID: 9bff9ef080d431aa
CSeq: 860568219 INVITE
Server: Asterisk PBX 12.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“kabejames.homelinux.com”, nonce="44c7a917"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘9bff9ef080d431aa’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:94.252.40.87:5060 —>
ACK sip:1066@kabejames.homelinux.com:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bKcbe1c2fd400edb4d7
Max-Forwards: 70
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061;tag=as164560ca
Call-ID: 9bff9ef080d431aa
CSeq: 860568219 ACK
User-Agent: Media5-fone/4.1.2.2653 iOS/7.1.2
Content-Length: 0

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

<— SIP read from UDP:94.252.40.87:5060 —>
INVITE sip:1066@kabejames.homelinux.com:5061 SIP/2.0
Accept: application/conference-info+xml, application/sdp, message/sipfrag, multipart/mixed
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bK5dfef2394115594b1
Max-Forwards: 70
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061
Call-ID: 9bff9ef080d431aa
CSeq: 860568220 INVITE
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
Allow-Events: refer
Authorization: Digest username=“1064”,realm=“kabejames.homelinux.com”,nonce=“44c7a917”,uri=“sip:1066@kabejames.homelinux.com:5061”,response=“d277c29e909bc6f9f97a147572e788ea”,algorithm=MD5
Contact: sip:1064@192.168.178.70:5060;audio
Supported: 100rel, replaces, sdp-anat, join
User-Agent: Media5-fone/4.1.2.2653 iOS/7.1.2
Content-Disposition: session
Content-Type: application/sdp
Content-Length: 306

v=0
o=- 3251547188403486068 3251547188403486069 IN IP4 192.168.178.70
s=m5
c=IN IP4 192.168.178.70
t=0 0
a=sendrecv
m=audio 10000 RTP/AVP 0 8 96 125
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:125 telephone-event/8000
a=fmtp:96 mode=20
a=fmtp:125 0-15
a=sendrecv
<------------->
— (17 headers 14 lines) —
Sending to 94.252.40.87:5060 (NAT)
Using INVITE request as basis request - 9bff9ef080d431aa
Found peer ‘1064’ for ‘1064’ from 94.252.40.87:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 125
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format iLBC for ID 96
Found audio description format telephone-event for ID 125
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.70:10000
Looking for 1066 in default (domain kabejames.homelinux.com)
list_route: route/path hop: sip:1064@192.168.178.70:5060

<— Transmitting (NAT) to 94.252.40.87:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bK5dfef2394115594b1;received=94.252.40.87;rport=5060
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061
Call-ID: 9bff9ef080d431aa
CSeq: 860568220 INVITE
Server: Asterisk PBX 12.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1066@192.168.178.56:5061
Content-Length: 0

<------------>
– Executing [1066@default:1] Dial(“SIP/1064-00000006”, “SIP/1066”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 14474
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.82.203.71:5060:
INVITE sip:1066@10.82.203.71:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.56:5061;branch=z9hG4bK25022384
Max-Forwards: 70
From: “1064” sip:1064@192.168.178.56:5061;tag=as6ebae36b
To: sip:1066@10.82.203.71:5060
Contact: sip:1064@192.168.178.56:5061
Call-ID: 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.7.1
Date: Mon, 15 Dec 2014 21:03:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 689

v=0
o=root 1223308320 1223308320 IN IP4 192.168.178.56
s=Asterisk PBX 12.7.1
c=IN IP4 192.168.178.56
t=0 0
m=audio 14474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:1e5c76df077a4d5758e668ba5e8f5815
a=ice-pwd:74ec46430b865052425b616e47b3a3af
a=candidate:Hc0a8b238 1 UDP 2130706431 192.168.178.56 14474 typ host
a=candidate:S5efc2857 1 UDP 1694498815 94.252.40.87 14474 typ srflx raddr 192.168.178.56 rport 14474
a=candidate:Hc0a8b238 2 UDP 2130706430 192.168.178.56 14475 typ host
a=candidate:S5efc2857 2 UDP 1694498814 94.252.40.87 14475 typ srflx raddr 192.168.178.56 rport 14475
a=sendrecv


-- Called SIP/1066

Retransmitting #1 (no NAT) to 10.82.203.71:5060:
INVITE sip:1066@10.82.203.71:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.56:5061;branch=z9hG4bK25022384
Max-Forwards: 70
From: “1064” sip:1064@192.168.178.56:5061;tag=as6ebae36b
To: sip:1066@10.82.203.71:5060
Contact: sip:1064@192.168.178.56:5061
Call-ID: 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.7.1
Date: Mon, 15 Dec 2014 21:03:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 689

v=0
o=root 1223308320 1223308320 IN IP4 192.168.178.56
s=Asterisk PBX 12.7.1
c=IN IP4 192.168.178.56
t=0 0
m=audio 14474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:1e5c76df077a4d5758e668ba5e8f5815
a=ice-pwd:74ec46430b865052425b616e47b3a3af
a=candidate:Hc0a8b238 1 UDP 2130706431 192.168.178.56 14474 typ host
a=candidate:S5efc2857 1 UDP 1694498815 94.252.40.87 14474 typ srflx raddr 192.168.178.56 rport 14474
a=candidate:Hc0a8b238 2 UDP 2130706430 192.168.178.56 14475 typ host
a=candidate:S5efc2857 2 UDP 1694498814 94.252.40.87 14475 typ srflx raddr 192.168.178.56 rport 14475
a=sendrecv


Retransmitting #2 (no NAT) to 10.82.203.71:5060:
INVITE sip:1066@10.82.203.71:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.56:5061;branch=z9hG4bK25022384
Max-Forwards: 70
From: “1064” sip:1064@192.168.178.56:5061;tag=as6ebae36b
To: sip:1066@10.82.203.71:5060
Contact: sip:1064@192.168.178.56:5061
Call-ID: 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.7.1
Date: Mon, 15 Dec 2014 21:03:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 689

v=0
o=root 1223308320 1223308320 IN IP4 192.168.178.56
s=Asterisk PBX 12.7.1
c=IN IP4 192.168.178.56
t=0 0
m=audio 14474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:1e5c76df077a4d5758e668ba5e8f5815
a=ice-pwd:74ec46430b865052425b616e47b3a3af
a=candidate:Hc0a8b238 1 UDP 2130706431 192.168.178.56 14474 typ host
a=candidate:S5efc2857 1 UDP 1694498815 94.252.40.87 14474 typ srflx raddr 192.168.178.56 rport 14474
a=candidate:Hc0a8b238 2 UDP 2130706430 192.168.178.56 14475 typ host
a=candidate:S5efc2857 2 UDP 1694498814 94.252.40.87 14475 typ srflx raddr 192.168.178.56 rport 14475
a=sendrecv


Retransmitting #3 (no NAT) to 10.82.203.71:5060:
INVITE sip:1066@10.82.203.71:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.56:5061;branch=z9hG4bK25022384
Max-Forwards: 70
From: “1064” sip:1064@192.168.178.56:5061;tag=as6ebae36b
To: sip:1066@10.82.203.71:5060
Contact: sip:1064@192.168.178.56:5061
Call-ID: 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.7.1
Date: Mon, 15 Dec 2014 21:03:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 689

v=0
o=root 1223308320 1223308320 IN IP4 192.168.178.56
s=Asterisk PBX 12.7.1
c=IN IP4 192.168.178.56
t=0 0
m=audio 14474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:1e5c76df077a4d5758e668ba5e8f5815
a=ice-pwd:74ec46430b865052425b616e47b3a3af
a=candidate:Hc0a8b238 1 UDP 2130706431 192.168.178.56 14474 typ host
a=candidate:S5efc2857 1 UDP 1694498815 94.252.40.87 14474 typ srflx raddr 192.168.178.56 rport 14474
a=candidate:Hc0a8b238 2 UDP 2130706430 192.168.178.56 14475 typ host
a=candidate:S5efc2857 2 UDP 1694498814 94.252.40.87 14475 typ srflx raddr 192.168.178.56 rport 14475
a=sendrecv


<— SIP read from UDP:185.40.62.230:5060 —>

<------------->
Retransmitting #4 (no NAT) to 10.82.203.71:5060:
INVITE sip:1066@10.82.203.71:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.56:5061;branch=z9hG4bK25022384
Max-Forwards: 70
From: “1064” sip:1064@192.168.178.56:5061;tag=as6ebae36b
To: sip:1066@10.82.203.71:5060
Contact: sip:1064@192.168.178.56:5061
Call-ID: 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.7.1
Date: Mon, 15 Dec 2014 21:03:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 689

v=0
o=root 1223308320 1223308320 IN IP4 192.168.178.56
s=Asterisk PBX 12.7.1
c=IN IP4 192.168.178.56
t=0 0
m=audio 14474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:1e5c76df077a4d5758e668ba5e8f5815
a=ice-pwd:74ec46430b865052425b616e47b3a3af
a=candidate:Hc0a8b238 1 UDP 2130706431 192.168.178.56 14474 typ host
a=candidate:S5efc2857 1 UDP 1694498815 94.252.40.87 14474 typ srflx raddr 192.168.178.56 rport 14474
a=candidate:Hc0a8b238 2 UDP 2130706430 192.168.178.56 14475 typ host
a=candidate:S5efc2857 2 UDP 1694498814 94.252.40.87 14475 typ srflx raddr 192.168.178.56 rport 14475
a=sendrecv


<— SIP read from UDP:94.252.40.87:5060 —>
CANCEL sip:1066@kabejames.homelinux.com:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bK5dfef2394115594b1
Max-Forwards: 70
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061
Call-ID: 9bff9ef080d431aa
CSeq: 860568220 CANCEL
User-Agent: Media5-fone/4.1.2.2653 iOS/7.1.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 94.252.40.87:5060 (NAT)

<— Reliably Transmitting (NAT) to 94.252.40.87:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bK5dfef2394115594b1;received=94.252.40.87;rport=5060
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061;tag=as24073fb6
Call-ID: 9bff9ef080d431aa
CSeq: 860568220 INVITE
Server: Asterisk PBX 12.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (NAT) to 94.252.40.87:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bK5dfef2394115594b1;received=94.252.40.87;rport=5060
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061;tag=as24073fb6
Call-ID: 9bff9ef080d431aa
CSeq: 860568220 CANCEL
Server: Asterisk PBX 12.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061’ in 32000 ms (Method: INVITE)
== Spawn extension (default, 1066, 1) exited non-zero on ‘SIP/1064-00000006’

<— SIP read from UDP:94.252.40.87:5060 —>
ACK sip:1066@kabejames.homelinux.com:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.70;branch=z9hG4bK5dfef2394115594b1
Max-Forwards: 70
From: “1064” sip:1064@kabejames.homelinux.com:5061;tag=aa69693952
To: sip:1066@kabejames.homelinux.com:5061;tag=as24073fb6
Call-ID: 9bff9ef080d431aa
CSeq: 860568220 ACK
User-Agent: Media5-fone/4.1.2.2653 iOS/7.1.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘9bff9ef080d431aa’ Method: ACK

*CLI> reloadRetransmitting #5 (no NAT) to 10.82.203.71:5060:
INVITE sip:1066@10.82.203.71:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.56:5061;branch=z9hG4bK25022384
Max-Forwards: 70
From: “1064” sip:1064@192.168.178.56:5061;tag=as6ebae36b
To: sip:1066@10.82.203.71:5060
Contact: sip:1064@192.168.178.56:5061
Call-ID: 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.7.1
Date: Mon, 15 Dec 2014 21:03:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 689

v=0
o=root 1223308320 1223308320 IN IP4 192.168.178.56
s=Asterisk PBX 12.7.1
c=IN IP4 192.168.178.56
t=0 0
m=audio 14474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:1e5c76df077a4d5758e668ba5e8f5815
a=ice-pwd:74ec46430b865052425b616e47b3a3af
a=candidate:Hc0a8b238 1 UDP 2130706431 192.168.178.56 14474 typ host
a=candidate:S5efc2857 1 UDP 1694498815 94.252.40.87 14474 typ srflx raddr 192.168.178.56 rport 14474
a=candidate:Hc0a8b238 2 UDP 2130706430 192.168.178.56 14475 typ host
a=candidate:S5efc2857 2 UDP 1694498814 94.252.40.87 14475 typ srflx raddr 192.168.178.56 rport 14475
a=sendrecv


  sip set debug off

SIP Debugging Disabled
*CLI> [Dec 15 22:04:26] WARNING[2205]: chan_sip.c:4098 retrans_pkt: Retransmission timeout reached on transmission 256cc22c234d818378d0fbaf6d858038@192.168.178.56:5061 for seqno 102 (Critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response