Code:--- Audio is at 10246 Adding codec ulaw to SDP 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-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-Type: application/sdp Content-Length: 292 v=0 o=root 665561663 665561663 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 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: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: 198813419 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-LHLE-8707918f-3db175f4 User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:91.121.129.23:5060 ---> INVITE 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> Content-Type: application/sdp CSeq: 198813420 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>;tag=as613b4868 Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6 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 156526349187 IN IP4 91.121.128.136 s=SIP Call c=IN IP4 91.121.128.136 t=0 0 m=audio 32142 RTP/AVP 0 8 18 101 b=AS:82 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no 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) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G729 for ID 18 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 <--- 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-BPIK-87079191-6277c4a6;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: 198813420 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 10246 Adding codec ulaw to SDP 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-BPIK-87079191-6277c4a6;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: 198813420 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: 292 v=0 o=root 665561663 665561664 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 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: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: 198813420 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-YYHU-87079197-2495cec1 User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:91.121.129.23:5060 ---> INVITE 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> Content-Type: application/sdp CSeq: 198813421 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>;tag=as613b4868 Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK User-Agent: Cirpack/v4.76 (gw_sip) Content-Length: 243 v=0 o=cp10 156526349186 156526349188 IN IP4 91.121.128.136 s=SIP Call c=IN IP4 91.121.128.136 t=0 0 m=audio 32142 RTP/AVP 8 101 b=AS:82 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> --- (13 headers 12 lines) --- Sending to 91.121.129.23:5060 (NAT) Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (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 <--- 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-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-Length: 0