SIP Debugging Enabled 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:2128@10.10.60.15 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK34cc3e15;rport From: "asterisk" ;tag=as1483c024 To: Contact: Call-ID: 0fd9db7607c615fd6fda8df021a4e134@10.10.10.1 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 10.10.60.15:5060 Scheduling destruction of call '0fd9db7607c615fd6fda8df021a4e134@10.10.10.1' in 15000 ms hunpbx*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK34cc3e15;rport From: "asterisk" ;tag=as1483c024 To: ;tag=7CDB5CAC-A6517F7F CSeq: 102 NOTIFY Call-ID: 0fd9db7607c615fd6fda8df021a4e134@10.10.10.1 Contact: Event: message-summary User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Content-Length: 0 10 headers, 0 lines Destroying call '0fd9db7607c615fd6fda8df021a4e134@10.10.10.1' hunpbx*CLI> Sip read: INVITE sip:2123@10.10.10.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK5da5171b880423E6 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: CSeq: 1 INVITE Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 1546011516 1546011516 IN IP4 10.10.60.11 s=Polycom IP Phone c=IN IP4 10.10.60.11 t=0 0 a=sendrecv m=audio 2256 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 14 headers, 11 lines Using latest request as basis request Sending to 10.10.60.11 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 10.10.60.11:0 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000c(ULAW|ALAW|H263), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK5da5171b880423E6 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as11a299b1 Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="613d152c" Content-Length: 0 to 10.10.60.11:5060 Scheduling destruction of call 'd1cbc7ff-6b17b1a9-e75472ba@10.10.60.11' in 15000 ms Found user '2124' hunpbx*CLI> Sip read: ACK sip:2123@10.10.10.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK5da5171b880423E6 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as11a299b1 CSeq: 1 ACK Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Max-Forwards: 70 Content-Length: 0 10 headers, 0 lines hunpbx*CLI> Sip read: INVITE sip:2123@10.10.10.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK6c50ae303235BD87 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: CSeq: 2 INVITE Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Supported: 100rel,timer,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="2124", realm="asterisk", nonce="613d152c", uri="sip:2123@10.10.10.1:5060", response="8ba9a7fbaf56d91800dcd8c8fbdf8f29", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 247 v=0 o=- 1546011516 1546011516 IN IP4 10.10.60.11 s=Polycom IP Phone c=IN IP4 10.10.60.11 t=0 0 a=sendrecv m=audio 2256 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 15 headers, 11 lines Using latest request as basis request Sending to 10.10.60.11 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer RTP is at port 10.10.60.11:0 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x8000c(ULAW|ALAW|H263), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user '2124' Looking for 2123 in default list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK6c50ae303235BD87 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as178be62f Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.10.60.11:5060 -- Executing Macro("SIP/2124-dde4", "oneline|SIP/2123") in new stack -- Executing Dial("SIP/2124-dde4", "SIP/2123|20|tr") in new stack We're at 10.10.10.1 port 17188 Answering/Requesting with root capability 4 Answering with preferred capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) 12 headers, 11 lines Reliably Transmitting: INVITE sip:2123@10.10.60.13 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK61578547;rport From: "Beth Beck" ;tag=as2ae01ee8 To: Contact: Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 22 Jul 2004 20:49:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp ontent-Length: 234 v=0 o=root 1279 1279 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 17188 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.10.60.13:5060 -- Called 2123 Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK6c50ae303235BD87 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as178be62f Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.10.60.11:5060 hunpbx*CLI> Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK61578547;rport From: "Beth Beck" ;tag=as2ae01ee8 To: ;tag=4E9650F7-C86DFEE4 CSeq: 102 INVITE Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Content-Length: 0 9 headers, 0 lines hunpbx*CLI> Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK61578547;rport From: "Beth Beck" ;tag=as2ae01ee8 To: ;tag=4E9650F7-C86DFEE4 CSeq: 102 INVITE Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Allow-Events: talk,hold,conference Content-Length: 0 10 headers, 0 lines -- SIP/2123-74c9 is ringing hunpbx*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK61578547;rport From: "Beth Beck" ;tag=as2ae01ee8 To: ;tag=4E9650F7-C86DFEE4 CSeq: 102 INVITE Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Content-Type: application/sdp Content-Length: 183 v=0pbx*CLI> o=- 461760469 461760469 IN IP4 10.10.60.13 s=Polycom IP Phone c=IN IP4 10.10.60.13 t=0 0 m=audio 2240 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 11 headers, 8 lines Found RTP audio format 0 Found RTP audio format 101 Peer RTP is at port 10.10.60.13:0 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000c(ULAW|ALAW|H263), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.60.13, port 5060 Transmitting: ACK sip:2123@10.10.60.13:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK4b7c851a;rport From: "Beth Beck" ;tag=as2ae01ee8 To: ;tag=4E9650F7-C86DFEE4 Contact: Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.10.60.13:5060 -- SIP/2123-74c9 answered SIP/2124-dde4 We're at 10.10.10.1 port 11604 Answering with preferred capability 0x4(ULAW) Answering with preferred capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK6c50ae303235BD87 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as178be62f Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=root 1279 1279 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 11604 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 10.10.60.11:5060 -- Attempting native bridge of SIP/2124-dde4 and SIP/2123-74c9 -- Attempting native bridge of SIP/2124-dde4 and SIP/2123-74c9 hunpbx*CLI> Sip read: ACK sip:2123@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK660d872458F44A2B From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as178be62f CSeq: 2 ACK Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Max-Forwards: 70 Content-Length: 0 10 headers, 0 lines Destroying call '1930c7c5-b751cf53-451788aa@10.10.60.15' hunpbx*CLI> Sip read: BYE sip:2124@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.60.13:5060;branch=z9hG4bKee840a2eBD0D0371 From: ;tag=4E9650F7-C86DFEE4 To: "Beth Beck" ;tag=as2ae01ee8 CSeq: 1 BYE Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Max-Forwards: 70 Content-Length: 0 hunpbx*CLI> 11 headers, 0 lines Sending to 10.10.60.13 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.60.13:5060;branch=z9hG4bKee840a2eBD0D0371 From: ;tag=4E9650F7-C86DFEE4 To: "Beth Beck" ;tag=as2ae01ee8 Call-ID: 1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.10.60.13:5060 == Spawn extension (macro-oneline, s, 1) exited non-zero on 'SIP/2124-dde4' in macro 'oneline' == Spawn extension (default, 2123, 1) exited non-zero on 'SIP/2124-dde4' set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.60.11, port 5060 Reliably Transmitting: BYE sip:2124@10.10.60.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK2d8dcecb;rport From: ;tag=as178be62f To: "Beth Beck" ;tag=DA043308-24A8B2A5 Contact: Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.10.60.11:5060 Destroying call '1e2f23d7450f36b36ceed0786efbc34d@10.10.10.1' hunpbx*CLI> Sip read: BYE sip:2123@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK3ec0ba35C9F32258 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as178be62f CSeq: 3 BYE Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Proxy-Authorization: Digest username="2124", realm="asterisk", nonce="613d152c", uri="sip:2123@10.10.10.1:5060", response="7de4bd931197ba9b1540ba8f85a86178", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 12 headers, 0 lines Sending to 10.10.60.11 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.60.11:5060;branch=z9hG4bK3ec0ba35C9F32258 From: "Beth Beck" ;tag=DA043308-24A8B2A5 To: ;tag=as178be62f Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Seq: 3 BYE> User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 10.10.60.11:5060 hunpbx*CLI> Sip read: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK2d8dcecb;rport From: ;tag=as178be62f To: "Beth Beck" ;tag=DA043308-24A8B2A5 CSeq: 102 BYE Call-ID: d1cbc7ff-6b17b1a9-e75472ba@10.10.60.11 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.2.0 Content-Length: 0 9 headers, 0 lines Message is BYE Destroying call 'd1cbc7ff-6b17b1a9-e75472ba@10.10.60.11' hunpbx*CLI> sip no deub No such command 'sip no deub' (type 'help' for help) 11 headers, 0 linesdeub Reliably Transmitting: OPTIONS sip:10.10.60.9 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1b148256;rport From: "asterisk" ;tag=as5c93b7a3 To: Contact: Call-ID: 00762b9d7015d8f44ca52846180211cd@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 22 Jul 2004 20:50:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.10.60.9:5060 11 headers, 0 linesdeub Reliably Transmitting: OPTIONS sip:10.10.60.15 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK0e5f8e49;rport From: "asterisk" ;tag=as6a44d81d To: Contact: Call-ID: 6944c7810164693711f0dfa358408ace@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 22 Jul 2004 20:50:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.10.60.15:5060 11 headers, 0 lines Reliably Transmitting: OPTIONS sip:10.10.60.2 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK32f60a6b;rport From: "asterisk" ;tag=as6f24b4df To: Contact: Call-ID: 5683bbeb2a2babc56ede82c756f9e788@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Thu, 22 Jul 2004 20:50:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 10.10.60.2:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK1b148256;rport From: "asterisk" ;tag=as5c93b7a3 To: ;tag=731DDA01-ACB80BD8 CSeq: 102 OPTIONS Call-ID: 00762b9d7015d8f44ca52846180211cd@10.10.10.1 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.1.0 Content-Length: 0 10 headers, 0 lines Destroying call '00762b9d7015d8f44ca52846180211cd@10.10.10.1'