asterisk>set verbose 4 asterisk>set debug 4 asterisk>sip debug Sip read: INVITE sip:0035929862508@12.34.56.78:65123 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK071b.8d316717.0 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;rport=5060;branch=z9hG4bKc0a8026f000000674382fd1a000003a7000001d1 Content-Length: 359 Contact: Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 Content-Type: application/sdp CSeq: 2 INVITE From: "unknown";tag=9556893014025 Max-Forwards: 16 To: User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digest username="123",realm="12.34.56.78",nonce="4382fe4c0d241c7728cc68fc25b74412706cec49",uri="sip:0035929862508@12.34.56.78",response="6127578e2da023aa7e0ac39e861325ab",cnonce="955690314543",qop="auth",nc="00000001" v=0 o=- 3341646746 3341646746 IN IP4 192.168.2.111 s=SJphone c=IN IP4 192.168.2.111 t=0 0 a=direction:active m=audio 49172 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 a=direction:active 14 headers, 16 lines Using latest request as basis request Sending to 12.34.56.78 : 5060 (NAT) Found peer 'pstn' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.111:49172 Found description format GSM Found description format iLBC Found description format iLBC Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 0035929862508 in pstn list_route: hop: list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK071b.8d316717.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd1a000003a7000001d1 From: "unknown";tag=9556893014025 To: Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 12.34.56.78:5060 Nov 22 13:12:33 WARNING[17952]: pbx.c:938 pbx_substitute_variables_temp: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' -- Executing Dial("SIP/12.34.56.78-08967378", "SIP/35929862508@83.88.15.24") in new stack We're at 12.34.56.78 port 18840 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:35929862508@83.88.15.24 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK03dedf5f From: "unknown" ;tag=as1df18db6 To: Contact: Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 22 Nov 2005 11:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17952 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 18840 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 83.88.15.24:5060 -- Called 35929862508@83.88.15.24 rb5*CLI> Sip read: SIP/2.0 100 Trying CSeq: 102 INVITE Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 From: "unknown";tag=as1df18db6 To: ;tag=3ef4af85-10b072 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK03dedf5f User-Agent: Quintum/1.0.0 Quintum: 0b0a32323738333937303035 8 headers, 0 lines rb5*CLI> Sip read: SIP/2.0 180 Ringing CSeq: 102 INVITE Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 From: "unknown";tag=as1df18db6 To: ;tag=3ef4af85-10b072 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK03dedf5f User-Agent: Quintum/1.0.0 7 headers, 0 lines -- SIP/83.88.15.24-290d is ringing Transmitting (NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK071b.8d316717.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd1a000003a7000001d1 From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER ontact: Content-Length: 0 to 12.34.56.78:5060 rb5*CLI> Sip read: SIP/2.0 200 OK CSeq: 102 INVITE Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 Contact: Content-Type: application/sdp From: "unknown";tag=as1df18db6 To: ;tag=3ef4af85-10b072 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK03dedf5f Content-Length: 167 User-Agent: Quintum/1.0.0 v=0 o=Quintum 3700 17952 IN IP4 83.88.15.24 s=VoipCall c=IN IP4 83.88.15.24 t=0 0 m=audio 12618 RTP/AVP 8 c=IN IP4 83.88.15.24 a=rtpmap:8 pcma/8000/1 10 headers, 8 lines Found RTP audio format 8 Peer audio RTP is at port 83.88.15.24:12618 Found description format pcma Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 83.88.15.24, port 5060 Transmitting: ACK sip:35929862508@83.88.15.24 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK4c97e34d From: "unknown" ;tag=as1df18db6 To: ;tag=3ef4af85-10b072 Contact: Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 (no NAT) to 83.88.15.24:5060 -- SIP/83.88.15.24-290d answered SIP/12.34.56.78-08967378 We're at 12.34.56.78 port 19172 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK071b.8d316717.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd1a000003a7000001d1 Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17952 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 -- Attempting native bridge of SIP/12.34.56.78-08967378 and SIP/83.88.15.24-290d rb5*CLI> Sip read: ACK sip:0035929862508@12.34.56.78:65123 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 12.34.56.78;branch=0 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;rport=5060;branch=z9hG4bKc0a8026f000000674382fd2000006cff000001d9 Content-Length: 0 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 2 ACK From: "unknown";tag=9556893014025 Max-Forwards: 16 To: ;tag=as0787ac71 User-Agent: SJphone/1.60.289a (SJ Labs) P-hint: rr-enforced 12 headers, 0 lines Nov 22 13:12:41 NOTICE[17952]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible rb5*CLI> Sip read: INVITE sip:0035929862508@12.34.56.78:65123 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;rport=5060;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Content-Length: 235 Contact: Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 Content-Type: application/sdp CSeq: 3 INVITE From: "unknown";tag=9556893014025 Max-Forwards: 16 To: ;tag=as0787ac71 User-Agent: SJphone/1.60.289a (SJ Labs) P-hint: rr-enforced v=0 o=- 3341646746 3341646747 IN IP4 192.168.2.111 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 49172 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 a=direction:active 14 headers, 11 lines Using latest request as basis request Sending to 12.34.56.78 : 5060 (NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:49172 Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) We're at 12.34.56.78 port 19172 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17953 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 rb5*CLI> Sip read: BYE sip:0035929862508@12.34.56.78:65123 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bKe61b.be0d7b33.0 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;rport=5060;branch=z9hG4bKc0a8026f000000674382fd2500001e4e000001de Content-Length: 0 Also: sip:00359898467099@12.34.56.78 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 4 BYE From: "unknown";tag=9556893014025 Max-Forwards: 16 To: ;tag=as0787ac71 User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digest username="123",realm="12.34.56.78",nonce="4382fe4c0d241c7728cc68fc25b74412706cec49",uri="sip:0035929862508@12.34.56.78",response="6127578e2da023aa7e0ac39e861325ab",cnonce="955690314543",qop="auth",nc="00000001" P-hint: rr-enforced 14 headers, 0 lines Sending to 12.34.56.78 : 5060 (NAT) Nov 22 13:12:44 NOTICE[17952]: chan_sip.c:7577 handle_request: Client '12.34.56.78' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead Looking for 00359898467099 in pstn Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bKe61b.be0d7b33.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd2500001e4e000001de Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 4 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 12.34.56.78:5060 Nov 22 13:12:44 WARNING[17952]: pbx.c:938 pbx_substitute_variables_temp: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' -- Executing Dial("SIP/83.88.15.24-290d", "SIP/359898467099@83.88.15.24") in new stack rb5*CLI> Sip read: ACK sip:0035929862508@12.34.56.78:65123 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 12.34.56.78;branch=0 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;rport=5060;branch=z9hG4bKc0a8026f000000674382fd250000629f000001dd Content-Length: 0 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 ACK From: "unknown";tag=9556893014025 Max-Forwards: 16 To: ;tag=as0787ac71 User-Agent: SJphone/1.60.289a (SJ Labs) P-hint: rr-enforced 12 headers, 0 lines We're at 12.34.56.78 port 19936 Answering/Requesting with root capability 0x8 (alaw) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting: INVITE sip:359898467099@83.88.15.24 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK67c785a3 From: "unknown" ;tag=as72310c82 To: Contact: Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 CSeq: 102 INVITE User-Agent: Asterisk Date: Tue, 22 Nov 2005 11:12:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 242 =05*CLI> o=root 17952 17952 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19936 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 83.88.15.24:5060 -- Called 359898467099@83.88.15.24 rb5*CLI> Sip read: SIP/2.0 100 Trying CSeq: 102 INVITE Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 From: "unknown";tag=as72310c82 To: ;tag=3ef4af85-10b075 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK67c785a3 User-Agent: Quintum/1.0.0 Quintum: 0b0a32323738333937303038 8 headers, 0 lines Retransmitting #1 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17953 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 Retransmitting #2 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17953 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 Retransmitting #3 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17953 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 Retransmitting #4 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17953 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 Retransmitting #5 (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 12.34.56.78;branch=z9hG4bK171b.3c2b7394.0;received=12.34.56.78;rport=5060 Via: SIP/2.0/UDP 192.168.2.111;received=85.187.165.42;branch=z9hG4bKc0a8026f000000674382fd250000772a000001da Record-Route: From: "unknown";tag=9556893014025 To: ;tag=as0787ac71 Call-ID: B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 17952 17953 IN IP4 12.34.56.78 s=session c=IN IP4 12.34.56.78 t=0 0 m=audio 19172 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 12.34.56.78:5060 rb5*CLI> Sip read: SIP/2.0 180 Ringing CSeq: 102 INVITE Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 From: "unknown";tag=as72310c82 To: ;tag=3ef4af85-10b075 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK67c785a3 User-Agent: Quintum/1.0.0 7 headers, 0 lines -- SIP/83.88.15.24-5a29 is ringing Nov 22 13:12:50 WARNING[17952]: chan_sip.c:696 retrans_pkt: Maximum retries exceeded on call B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111 for seqno 3 (Non-critical Response) Destroying call 'B80A0402-3449-4537-B762-E69BB23572AA@192.168.2.111' rb5*CLI> Sip read: SIP/2.0 200 OK CSeq: 102 INVITE Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 Contact: Content-Type: application/sdp From: "unknown";tag=as72310c82 To: ;tag=3ef4af85-10b075 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK67c785a3 Content-Length: 167 User-Agent: Quintum/1.0.0 v=0 o=Quintum 3706 17952 IN IP4 83.88.15.24 s=VoipCall c=IN IP4 83.88.15.24 t=0 0 m=audio 12630 RTP/AVP 8 c=IN IP4 83.88.15.24 a=rtpmap:8 pcma/8000/1 10 headers, 8 lines Found RTP audio format 8 Peer audio RTP is at port 83.88.15.24:12630 Found description format pcma Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 83.88.15.24, port 5060 Transmitting: ACK sip:359898467099@83.88.15.24 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK57dddea1 From: "unknown" ;tag=as72310c82 To: ;tag=3ef4af85-10b075 Contact: Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 (no NAT) to 83.88.15.24:5060 -- SIP/83.88.15.24-5a29 answered SIP/83.88.15.24-290d -- Attempting native bridge of SIP/83.88.15.24-290d and SIP/83.88.15.24-5a29 Nov 22 13:12:52 NOTICE[17952]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible rb5*CLI> Sip read: BYE sip:123@12.34.56.78:65123 SIP/2.0 CSeq: 103 BYE Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 From: ;tag=3ef4af85-10b075 To: "unknown";tag=as72310c82 Via: SIP/2.0/UDP 83.88.15.24;branch=z9hG4bK-tenor-3ef4-af85-7eebe User-Agent: Quintum/1.0.0 Max-Forwards: 70 8 headers, 0 lines Sending to 83.88.15.24 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 83.88.15.24;branch=z9hG4bK-tenor-3ef4-af85-7eebe From: ;tag=3ef4af85-10b075 To: "unknown";tag=as72310c82 Call-ID: 5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78 CSeq: 103 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 83.88.15.24:5060 set_destination: Parsing for address/port to send to set_destination: set destination to 83.88.15.24, port 5060 Reliably Transmitting: BYE sip:35929862508@83.88.15.24 SIP/2.0 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK46e8287f From: "unknown" ;tag=as1df18db6 To: ;tag=3ef4af85-10b072 Contact: Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 (no NAT) to 83.88.15.24:5060 rb5*CLI> Sip read: SIP/2.0 200 OK CSeq: 103 BYE Call-ID: 1adc635902bae332284b12ae3c2b9bbb@12.34.56.78 From: "unknown";tag=as1df18db6 To: ;tag=3ef4af85-10b072 Via: SIP/2.0/UDP 12.34.56.78:65123;branch=z9hG4bK46e8287f 6 headers, 0 lines Destroying call '5a4c8f4164bc7ef0084b76af31fd665f@12.34.56.78' Destroying call '1adc635902bae332284b12ae3c2b9bbb@12.34.56.78' In /var/log/asterisk/messages can be seen the following message: Nov 22 13:11:50 WARNING[17952]: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' Nov 22 13:12:02 NOTICE[17952]: RFC3389 support incomplete. Turn off on client if possible Nov 22 13:12:08 NOTICE[17952]: Client '12.34.56.78' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead Nov 22 13:12:08 WARNING[17952]: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' Nov 22 13:12:16 NOTICE[17952]: RFC3389 support incomplete. Turn off on client if possible