<------------> set_destination: Parsing for address/port to send to set_destination: set destination to 87.248.56.100:5060 Audio is at 10028 Adding codec 100008 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 87.248.56.100:5060: INVITE sip:+390633040237@87.248.56.100 SIP/2.0 Via: SIP/2.0/UDP 87.248.56.102:5060;branch=z9hG4bK2ab7dbb7 Max-Forwards: 70 From: ;tag=as77930702 To: ;tag=3f8d7cfe0000be8c Contact: Call-ID: 04eca337110c4dbc1c8f2fd439bcc2d2@87.248.56.102:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 12.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 276 v=0 o=root 1953170441 1953170444 IN IP4 87.248.56.102 s=Asterisk PBX 12.4.0 c=IN IP4 87.248.56.102 t=0 0 m=audio 10028 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv