<--- SIP read from 192.114.XX.YY:13491 ---> INVITE sip:039372919@192.AAA.BBB.CCC:5060;user=phone SIP/2.0 Call-ID: 6998677383823422845-1234193639@192.114.XX.YY From: ;tag=27529 To: Content-Type: application/sdp CSeq: 1 INVITE Via: SIP/2.0/UDP 192.114.XX.YY:5060;branch=z9hG4bK-61204c00005af57d-c0724506-1 Contact: sip:0@192.114.XX.YY:5060;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO Supported: timer,100rel Max-Forwards: 70 Content-Length: 204 v=0 o=MG4000|2.0 41 108 IN IP4 192.115.ZZ.WW s=- c=IN IP4 192.115.ZZ.WW t=0 0 m=audio 34804 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:10 a=rtpmap:13 CN/8000 <-------------> --- (12 headers 10 lines) --- Sending to 192.114.XX.YY : 5060 (no NAT) Using INVITE request as basis request - 6998677383823422845-1234193639@192.114.XX.YY Found peer 'bezeqint0' Found RTP audio format 8 Found RTP audio format 101 Found RTP audio format 13 Peer audio RTP is at port 192.115.ZZ.WW:34804 Found audio description format telephone-event for ID 101 Found audio description format CN for ID 13 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x3 (telephone-event|CN), combined - 0x0 (nothing) Peer audio RTP is at port 192.115.ZZ.WW:34804 Looking for 039370000 in bezeqint_inbound (domain 192.AAA.BBB.CCC) list_route: hop: localhost*CLI> <--- Transmitting (no NAT) to 192.114.XX.YY:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.114.XX.YY:5060;branch=z9hG4bK-61204c00005af57d-c0724506-1;received=192.114.XX.YY From: ;tag=27529 To: Call-ID: 6998677383823422845-1234193639@192.114.XX.YY CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> Disconnected from Asterisk server Asterisk ended with exit status 127