<-- SIP read from 83.97.27.7:6108: INVITE sip:990012028447171@212.21.135.250 SIP/2.0 To: From: 212.21.135.250;tag=510a7807 Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-603223729-1--d87543-;rport Call-ID: bb02984e803c1e6d CSeq: 1 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 233 v=0 o=- 18460598 18460613 IN IP4 192.168.29.3 s=eyeBeam c=IN IP4 192.168.29.3 t=0 0 m=audio 8504 RTP/AVP 18 8 101 a=alt:1 1 : 942AED38 A84C5074 192.168.29.3 8504 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (12 headers 10 lines)--- Using INVITE request as basis request - bb02984e803c1e6d Sending to 192.168.29.3 : 6108 (NAT) Reliably Transmitting (NAT) to 83.97.27.7:6108: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-603223729-1--d87543-;received=83.97.27.7;rport=6108 From: 212.21.135.250;tag=510a7807 To: ;tag=as1b9a0242 Call-ID: bb02984e803c1e6d CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="58b24a40" Content-Length: 0 --- Scheduling destruction of call 'bb02984e803c1e6d' in 15000 ms Found user 'u333' asterisk*CLI> <-- SIP read from 83.97.27.7:6108: ACK sip:990012028447171@212.21.135.250 SIP/2.0 To: ;tag=as1b9a0242 From: 212.21.135.250;tag=510a7807 Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-603223729-1--d87543-;rport Call-ID: bb02984e803c1e6d CSeq: 1 ACK Content-Length: 0 --- (7 headers 0 lines)--- asterisk*CLI> <-- SIP read from 83.97.27.7:6108: INVITE sip:990012028447171@212.21.135.250 SIP/2.0 To: From: 212.21.135.250;tag=510a7807 Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-28665091-1--d87543-;rport Call-ID: bb02984e803c1e6d CSeq: 2 INVITE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="u333",realm="asterisk",nonce="58b24a40",uri="sip:990012028447171@212.21.135.250",response="24fe3fd9fc91742bd91911b516b5b5ed",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 233 v=0 o=- 18460598 18460613 IN IP4 192.168.29.3 s=eyeBeam c=IN IP4 192.168.29.3 t=0 0 m=audio 8504 RTP/AVP 18 8 101 a=alt:1 1 : 942AED38 A84C5074 192.168.29.3 8504 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 10 lines)--- Using INVITE request as basis request - bb02984e803c1e6d Sending to 192.168.29.3 : 6108 (NAT) Found user 'u333' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.29.3:8504 Found description format telephone-event Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - (g729|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 990012028447171 in internal (domain 212.21.135.250) list_route: hop: Transmitting (NAT) to 83.97.27.7:6108: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-28665091-1--d87543-;received=83.97.27.7;rport=6108 From: 212.21.135.250;tag=510a7807 To: Call-ID: bb02984e803c1e6d CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/u333-2274", "CALLERID(number)=35929714904") in new stack -- Executing Dial("SIP/u333-2274", "OH323/0012028447171@212.39.70.49|60|T") in new stack -- H.323 call to 0012028447171@212.39.70.49 with codec(s) alaw g729 ulaw -- Outbound H.323 call to destination '0012028447171@212.39.70.49', channel 'OH323/0012028447171@212.39.70.49-8b4de24'. -- Called 0012028447171@212.39.70.49 Destroying call '12cca8c0-13c4-99-25d28-693b' > H.323 call 'ip$localhost/6512-08b4de24', exception CALL_PROGRESS. -- OH323/0012028447171@212.39.70.49-8b4de24 is making progress passing it to SIP/u333-2274 We're at 212.21.135.250 port 17240 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (NAT) to 83.97.27.7:6108: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-28665091-1--d87543-;received=83.97.27.7;rport=6108 From: 212.21.135.250;tag=510a7807 To: ;tag=as2128dd3e Call-ID: bb02984e803c1e6d CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 760 760 IN IP4 212.21.135.250 s=session c=IN IP4 212.21.135.250 t=0 0 m=audio 17240 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- > H.323 call 'ip$localhost/6512-08b4de24', exception CALL_ESTABLISHED. -- OH323/0012028447171@212.39.70.49-8b4de24 answered SIP/u333-2274 We're at 212.21.135.250 port 17240 Adding codec 0x100 (g729) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 83.97.27.7:6108: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-28665091-1--d87543-;received=83.97.27.7;rport=6108 From: 212.21.135.250;tag=510a7807 To: ;tag=as2128dd3e Call-ID: bb02984e803c1e6d CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 760 761 IN IP4 212.21.135.250 s=session c=IN IP4 212.21.135.250 t=0 0 m=audio 17240 RTP/AVP 18 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk*CLI> <-- SIP read from 83.97.27.7:6108: ACK sip:990012028447171@212.21.135.250 SIP/2.0 To: ;tag=as2128dd3e From: 212.21.135.250;tag=510a7807 Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-376209151-1--d87543-;rport Call-ID: bb02984e803c1e6d CSeq: 2 ACK Contact: Max-Forwards: 70 Proxy-Authorization: Digest username="u333",realm="asterisk",nonce="58b24a40",uri="sip:990012028447171@212.21.135.250",response="24fe3fd9fc91742bd91911b516b5b5ed",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 <-- SIP read from 83.97.27.7:6108: BYE sip:990012028447171@212.21.135.250 SIP/2.0 To: ;tag=as2128dd3e From: 212.21.135.250;tag=510a7807 Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-657465580-1--d87543-;rport Call-ID: bb02984e803c1e6d CSeq: 3 BYE Contact: Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Proxy-Authorization: Digest username="u333",realm="asterisk",nonce="58b24a40",uri="sip:990012028447171@212.21.135.250",response="a7df9a5ab52a754a60cebb4398fb349e",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 --- (12 headers 0 lines)--- Sending to 192.168.29.3 : 6108 (NAT) Transmitting (NAT) to 83.97.27.7:6108: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.29.3:6108;branch=z9hG4bK-d87543-657465580-1--d87543-;received=83.97.27.7;rport=6108 From: 212.21.135.250;tag=510a7807 To: ;tag=as2128dd3e Call-ID: bb02984e803c1e6d CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- -- Hungup 'OH323/0012028447171@212.39.70.49-8b4de24' -- H.323 call 'ip$localhost/6512-08b4de24' cleared, reason 1 (Cleared by local user), established (5 sec) Destroying call 'bb02984e803c1e6d' asterisk*CLI> asterisk*CLI>