<------------>
set_destination: Parsing <sip:+390633040237@87.248.56.100> 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: <sip:0287250030@87.248.56.102>;tag=as77930702
To: <sip:+390633040237@87.248.56.100;user=phone>;tag=3f8d7cfe0000be8c
Contact: <sip:0287250030@87.248.56.102:5060>
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