Code:
<------------>
Audio is at 10246
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813421 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 665561663 665561665 IN IP4 185.246.18.202
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 185.246.18.202
t=0 0
m=audio 10246 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:91.121.129.23:5060 --->
ACK sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
CSeq: 198813421 ACK
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-SAQW-8707921f-562ea85e
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

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

<--- SIP read from UDP:192.168.130.49:5060 --->
BYE sip:0383723596@192.168.130.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565
From: <sip:214@192.168.130.49:5060>;tag=2669272485
To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 2 BYE
Contact: <sip:214@192.168.130.49:5060>
Max-Forwards: 70
User-Agent: Yealink SIP-T41S 66.84.0.15
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.130.49:5060 (NAT)
Scheduling destruction of SIP dialog '56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.130.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565;received=192.168.130.49;rport=5060
From: <sip:214@192.168.130.49:5060>;tag=2669272485
To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 2 BYE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 91.121.129.23:5060:
BYE sip:10.7.1.65:5060 SIP/2.0
Via: SIP/2.0/UDP 185.246.18.202:5060;branch=z9hG4bK26126eb9;rport
Route: <sip:91.121.129.23:5060;lr>
Max-Forwards: 70
From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:91.121.129.23:5060 --->
SIP/2.0 200 OK
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 102 BYE
From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Record-Route: <sip:91.121.129.23:5060;transport=udp;lr>;session=443934
To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Via: SIP/2.0/UDP 192.168.2.100:5060;received=192.168.2.100;rport=5060;branch=z9hG4bK26126eb9
Server: Cirpack/v4.76 (gw_sip)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' Method: ACK