Code:
<--- SIP read from UDP:91.121.129.23:5060 --->
INVITE sip:s@192.168.2.100:5060;transport=udp SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
Content-Type: application/sdp
CSeq: 198813419 INVITE
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
To: <sip:0383720044@10.7.1.65;user=phone>
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 318

v=0
o=cp10 156526349186 156526349186 IN IP4 91.121.128.136
s=SIP Call
c=IN IP4 91.121.128.136
t=0 0
m=audio 32142 RTP/AVP 8 18 0 101
b=AS:82
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 91.121.129.23:5060 (NAT)
Sending to 91.121.129.23:5060 (NAT)
Using INVITE request as basis request - 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Found peer 'trunk-ovh' for '0383723596' from 91.121.129.23:5060
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/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 91.121.128.136:32142
Looking for s in from-pstn (domain 192.168.2.100)
sip_route_dump: route/path hop: <sip:91.121.129.23:5060;lr>

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;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>
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813419 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-Length: 0


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

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;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: 198813419 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-Length: 0


<------------>
Audio is at 11998
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.130.49:5060:
INVITE sip:214@192.168.130.49:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport
Max-Forwards: 70
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>
Contact: <sip:0383723596@192.168.130.254:5060>
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Date: Thu, 08 Aug 2019 11:24:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 638829780 638829780 IN IP4 192.168.130.254
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 192.168.130.254
t=0 0
m=audio 11998 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
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:192.168.130.49:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
User-Agent: Yealink SIP-T41S 66.84.0.15
Content-Length: 0

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

<--- SIP read from UDP:192.168.130.49:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>;tag=2669272485
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
Contact: <sip:214@192.168.130.49:5060>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T41S 66.84.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:214@192.168.130.49:5060>

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-TKLX-87078c4d-1549530f;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: 198813419 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-Length: 0


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

<--- SIP read from UDP:192.168.130.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK4374ac72;rport=5060
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>;tag=2669272485
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 INVITE
Contact: <sip:214@192.168.130.49:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T41S 66.84.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 215

v=0
o=- 20037 20037 IN IP4 192.168.130.49
s=SDP data
c=IN IP4 192.168.130.49
t=0 0
m=audio 12244 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 11 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 - (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 192.168.130.49:12244
sip_route_dump: route/path hop: <sip:214@192.168.130.49:5060>
Transmitting (NAT) to 192.168.130.49:5060:
ACK sip:214@192.168.130.49:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.254:5060;branch=z9hG4bK1c2f34b1;rport
Max-Forwards: 70
From: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
To: <sip:214@192.168.130.49:5060>;tag=2669272485
Contact: <sip:0383723596@192.168.130.254:5060>
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Content-Length: 0