Jan 27 13:21:55 VERBOSE[17464]: Sip read: INVITE sip:104@pbx:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bKc8ca095f6429E136 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: CSeq: 1 INVITE Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.1 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 244 v=0 o=- 1106857063 1106857063 IN IP4 199.104.120.153 s=Polycom IP Phone c=IN IP4 199.104.120.153 t=0 0 m=audio 65504 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 Jan 27 13:21:55 VERBOSE[17464]: 14 headers, 10 lines Jan 27 13:21:55 VERBOSE[17464]: Using latest request as basis request Jan 27 13:21:55 VERBOSE[17464]: Sending to 199.104.120.153 : 5064 (non-NAT) Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 8 Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 18 Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:21:55 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.153:65504 Jan 27 13:21:55 DEBUG[17464]: Peer audio RTP is at port 199.104.120.153:65504 Jan 27 13:21:55 VERBOSE[17464]: Found description format PCMU Jan 27 13:21:55 VERBOSE[17464]: Found description format PCMA Jan 27 13:21:55 VERBOSE[17464]: Found description format G729 Jan 27 13:21:55 VERBOSE[17464]: Found description format telephone-event Jan 27 13:21:55 VERBOSE[17464]: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:21:55 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:21:55 DEBUG[17464]: Setting NAT on RTP to 0 Jan 27 13:21:55 VERBOSE[17464]: Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bKc8ca095f6429E136 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: ;tag=as3bec6958 Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.xmission.com", nonce="573a9714" Content-Length: 0 to 199.104.120.153:5064 Jan 27 13:21:55 VERBOSE[17464]: Scheduling destruction of call 'b48d1759-85b9503-3000e7b8@199.104.120.153' in 15000 ms Jan 27 13:21:55 VERBOSE[17464]: Found user '139' Jan 27 13:21:55 VERBOSE[17464]: Sip read: ACK sip:104@pbx:5061 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bKc8ca095f6429E136 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: ;tag=as3bec6958 CSeq: 1 ACK Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.1 Max-Forwards: 70 Content-Length: 0 Jan 27 13:21:55 VERBOSE[17464]: 11 headers, 0 lines Jan 27 13:21:55 DEBUG[17464]: Stopping retransmission on 'b48d1759-85b9503-3000e7b8@199.104.120.153' of Response 1: Found Jan 27 13:21:55 VERBOSE[17464]: Sip read: INVITE sip:104@pbx:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bK4ba2018c85202567 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: CSeq: 2 INVITE Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.1 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="139", realm="pbx.xmission.com", nonce="573a9714", uri="sip:104@pbx:5061;user=phone", response="115ae403cb4856691ed25255b2870fab", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 244 v=0 o=- 1106857063 1106857063 IN IP4 199.104.120.153 s=Polycom IP Phone c=IN IP4 199.104.120.153 t=0 0 m=audio 65504 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 Jan 27 13:21:55 VERBOSE[17464]: 15 headers, 10 lines Jan 27 13:21:55 VERBOSE[17464]: Using latest request as basis request Jan 27 13:21:55 VERBOSE[17464]: Sending to 199.104.120.153 : 5064 (non-NAT) Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 8 Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 18 Jan 27 13:21:55 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:21:55 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.153:65504 Jan 27 13:21:55 DEBUG[17464]: Peer audio RTP is at port 199.104.120.153:65504 Jan 27 13:21:55 VERBOSE[17464]: Found description format PCMU Jan 27 13:21:55 VERBOSE[17464]: Found description format PCMA Jan 27 13:21:55 VERBOSE[17464]: Found description format G729 Jan 27 13:21:55 VERBOSE[17464]: Found description format telephone-event Jan 27 13:21:55 VERBOSE[17464]: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:21:55 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:21:55 DEBUG[17464]: Setting NAT on RTP to 0 Jan 27 13:21:55 VERBOSE[17464]: Found user '139' Jan 27 13:21:55 DEBUG[17464]: Check for res for 139 Jan 27 13:21:55 DEBUG[17464]: Call from user '139' is 1 out of 0 Jan 27 13:21:55 VERBOSE[17464]: Looking for 104 in phone_admin Jan 27 13:21:55 DEBUG[17464]: build_route: Contact hop: Jan 27 13:21:55 VERBOSE[17464]: list_route: hop: Jan 27 13:21:55 VERBOSE[17464]: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bK4ba2018c85202567 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: ;tag=as11d583bc Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 199.104.120.153:5064 Jan 27 13:21:55 VERBOSE[17464]: -- Executing Macro("SIP/139-8369", "stdexten|104|hangup|pbx") in new stack Jan 27 13:21:55 VERBOSE[17464]: -- Executing DBget("SIP/139-8369", "chan=EXT/104") in new stack Jan 27 13:21:55 VERBOSE[17464]: -- DBget: varname=chan, family=EXT, key=104 Jan 27 13:21:55 VERBOSE[17464]: -- DBget: set variable chan to SIP/104 Jan 27 13:21:55 VERBOSE[17464]: -- Executing GotoIf("SIP/139-8369", "?6") in new stack Jan 27 13:21:55 DEBUG[17464]: Not taking any branch Jan 27 13:21:55 VERBOSE[17464]: -- Executing DBget("SIP/139-8369", "cid_name=EXT/139/cid_name") in new stack Jan 27 13:21:55 VERBOSE[17464]: -- DBget: varname=cid_name, family=EXT, key=139/cid_name Jan 27 13:21:55 VERBOSE[17464]: -- DBget: set variable cid_name to kevin Jan 27 13:21:55 VERBOSE[17464]: -- Executing SetCIDName("SIP/139-8369", "kevin") in new stack Jan 27 13:21:55 VERBOSE[17464]: -- Executing SetVar("SIP/139-8369", "_ALERT_INFO=none") in new stack Jan 27 13:21:55 VERBOSE[17464]: -- Executing Dial("SIP/139-8369", "SIP/104|25") in new stack Jan 27 13:21:55 DEBUG[17464]: Setting NAT on RTP to 0 Jan 27 13:21:55 DEBUG[17464]: Outgoing Call for 104 Jan 27 13:21:55 DEBUG[17464]: Call from user '104' is 1 out of 0 Jan 27 13:21:55 VERBOSE[17464]: We're at 199.104.120.160 port 19974 Jan 27 13:21:55 VERBOSE[17464]: Answering/Requesting with root capability 0x4 (ulaw) Jan 27 13:21:55 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:21:55 VERBOSE[17464]: 13 headers, 10 lines Jan 27 13:21:55 VERBOSE[17464]: Reliably Transmitting: INVITE sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK63036bca From: "kevin" ;tag=as40143110 To: Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 27 Jan 2005 20:21:55 GMT Alert-Info: none Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 222 v=0 o=root 17464 17464 IN IP4 199.104.120.160 s=session c=IN IP4 199.104.120.160 t=0 0 m=audio 19974 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 199.104.120.151:5060 Jan 27 13:21:55 VERBOSE[17464]: -- Called 104 Jan 27 13:21:55 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:55 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:55 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:55 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:55 VERBOSE[17464]: Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK63036bca From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 CSeq: 102 INVITE Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Length: 0 Jan 27 13:21:55 VERBOSE[17464]: 9 headers, 0 lines Jan 27 13:21:55 DEBUG[17464]: (Provisional) Stopping retransmission (but retaining packet) on '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' Request 102: Found Jan 27 13:21:55 VERBOSE[17464]: Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK63036bca From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 CSeq: 102 INVITE Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Allow-Events: talk,hold,conference Content-Length: 0 Jan 27 13:21:55 VERBOSE[17464]: 10 headers, 0 lines Jan 27 13:21:55 DEBUG[17464]: (Provisional) Stopping retransmission (but retaining packet) on '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' Request 102: Found Jan 27 13:21:55 VERBOSE[17464]: -- SIP/104-5a59 is ringing Jan 27 13:21:55 VERBOSE[17464]: Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bK4ba2018c85202567 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: ;tag=as11d583bc Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 199.104.120.153:5064 Jan 27 13:21:57 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK63036bca From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 CSeq: 102 INVITE Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Type: application/sdp Content-Length: 193 v=0 o=- 1106855047 1106855047 IN IP4 199.104.120.151 s=Polycom IP Phone c=IN IP4 199.104.120.151 t=0 0 m=audio 2222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:21:57 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:21:57 DEBUG[17464]: Acked pending invite 102 Jan 27 13:21:57 DEBUG[17464]: Stopping retransmission on '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' of Request 102: Found Jan 27 13:21:57 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:21:57 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:21:57 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.151:2222 Jan 27 13:21:57 DEBUG[17464]: Peer audio RTP is at port 199.104.120.151:2222 Jan 27 13:21:57 VERBOSE[17464]: Found description format PCMU Jan 27 13:21:57 VERBOSE[17464]: Found description format telephone-event Jan 27 13:21:57 VERBOSE[17464]: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:21:57 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:21:57 DEBUG[17464]: build_route: Contact hop: Jan 27 13:21:57 VERBOSE[17464]: list_route: hop: Jan 27 13:21:57 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:21:57 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:21:57 VERBOSE[17464]: Transmitting: ACK sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK1abbef3f From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:21:57 VERBOSE[17464]: -- SIP/104-5a59 answered SIP/139-8369 Jan 27 13:21:57 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:57 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:57 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:57 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:21:57 VERBOSE[17464]: We're at 199.104.120.160 port 15440 Jan 27 13:21:57 VERBOSE[17464]: Answering with preferred capability 0x4 (ulaw) Jan 27 13:21:57 VERBOSE[17464]: Answering with preferred capability 0x2 (gsm) Jan 27 13:21:57 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:21:57 VERBOSE[17464]: Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bK4ba2018c85202567 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: ;tag=as11d583bc Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 245 v=0 o=root 17464 17464 IN IP4 199.104.120.160 s=session c=IN IP4 199.104.120.160 t=0 0 m=audio 15440 RTP/AVP 0 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 199.104.120.153:5064 Jan 27 13:21:57 VERBOSE[17464]: -- Attempting native bridge of SIP/139-8369 and SIP/104-5a59 Jan 27 13:21:57 DEBUG[17464]: Deferring reinvite on 'b48d1759-85b9503-3000e7b8@199.104.120.153' Jan 27 13:21:57 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:21:57 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:21:57 VERBOSE[17464]: We're at 199.104.120.160 port 19974 Jan 27 13:21:57 VERBOSE[17464]: Answering/Requesting with root capability 0x4 (ulaw) Jan 27 13:21:57 VERBOSE[17464]: Answering with capability 0x8 (alaw) Jan 27 13:21:57 VERBOSE[17464]: Answering with capability 0x100 (g729) Jan 27 13:21:57 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:21:57 VERBOSE[17464]: 11 headers, 12 lines Jan 27 13:21:57 VERBOSE[17464]: Reliably Transmitting: INVITE sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK03232c80 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 272 v=0 o=root 17464 17465 IN IP4 199.104.120.153 s=session c=IN IP4 199.104.120.153 t=0 0 m=audio 65504 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 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 199.104.120.151:5060 Jan 27 13:21:57 VERBOSE[17464]: Sip read: ACK sip:104@199.104.120.160:5061 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.153:5064;branch=z9hG4bK85a1be8b9F7BCBD2 From: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA To: ;tag=as11d583bc CSeq: 2 ACK Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.1 Max-Forwards: 70 Content-Length: 0 Jan 27 13:21:57 VERBOSE[17464]: 11 headers, 0 lines Jan 27 13:21:57 DEBUG[17464]: Stopping retransmission on 'b48d1759-85b9503-3000e7b8@199.104.120.153' of Response 2: Found Jan 27 13:21:57 DEBUG[17464]: Sending pending reinvite on 'b48d1759-85b9503-3000e7b8@199.104.120.153' Jan 27 13:21:57 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:21:57 VERBOSE[17464]: set_destination: set destination to 199.104.120.153, port 5064 Jan 27 13:21:57 VERBOSE[17464]: We're at 199.104.120.160 port 15440 Jan 27 13:21:57 VERBOSE[17464]: Answering with preferred capability 0x4 (ulaw) Jan 27 13:21:57 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:21:57 VERBOSE[17464]: 11 headers, 10 lines Jan 27 13:21:57 VERBOSE[17464]: Reliably Transmitting: INVITE sip:139@199.104.120.153:5064 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK2cb16318;rport From: ;tag=as11d583bc To: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA Contact: Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 221 v=0 o=root 17464 17465 IN IP4 199.104.120.151 s=session c=IN IP4 199.104.120.151 t=0 0 m=audio 2222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 199.104.120.153:5064 Jan 27 13:21:57 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK03232c80 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 CSeq: 103 INVITE Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Type: application/sdp Content-Length: 193 v=0 o=- 1106855048 1106855048 IN IP4 199.104.120.151 s=Polycom IP Phone c=IN IP4 199.104.120.151 t=0 0 m=audio 2222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:21:57 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:21:57 DEBUG[17464]: Acked pending invite 103 Jan 27 13:21:57 DEBUG[17464]: Stopping retransmission on '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' of Request 103: Found Jan 27 13:21:57 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:21:57 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:21:57 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.151:2222 Jan 27 13:21:57 DEBUG[17464]: Peer audio RTP is at port 199.104.120.151:2222 Jan 27 13:21:57 VERBOSE[17464]: Found description format PCMU Jan 27 13:21:57 VERBOSE[17464]: Found description format telephone-event Jan 27 13:21:57 VERBOSE[17464]: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:21:57 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:21:57 DEBUG[17464]: build_route: Retaining previous route: Jan 27 13:21:57 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:21:57 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:21:57 VERBOSE[17464]: Transmitting: ACK sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK68b7814b From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:21:57 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK2cb16318;rport From: ;tag=as11d583bc To: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA CSeq: 102 INVITE Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.1 Content-Type: application/sdp Content-Length: 194 v=0 o=- 1106857064 1106857064 IN IP4 199.104.120.153 s=Polycom IP Phone c=IN IP4 199.104.120.153 t=0 0 m=audio 65504 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:21:57 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:21:57 DEBUG[17464]: Acked pending invite 102 Jan 27 13:21:57 DEBUG[17464]: Stopping retransmission on 'b48d1759-85b9503-3000e7b8@199.104.120.153' of Request 102: Found Jan 27 13:21:57 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:21:57 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:21:57 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.153:65504 Jan 27 13:21:57 DEBUG[17464]: Peer audio RTP is at port 199.104.120.153:65504 Jan 27 13:21:57 VERBOSE[17464]: Found description format PCMU Jan 27 13:21:57 VERBOSE[17464]: Found description format telephone-event Jan 27 13:21:57 VERBOSE[17464]: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:21:57 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:21:57 DEBUG[17464]: build_route: Contact hop: Jan 27 13:21:57 VERBOSE[17464]: list_route: hop: Jan 27 13:21:57 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:21:57 VERBOSE[17464]: set_destination: set destination to 199.104.120.153, port 5064 Jan 27 13:21:57 VERBOSE[17464]: Transmitting: ACK sip:139@199.104.120.153:5064 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK1c309296;rport From: ;tag=as11d583bc To: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA Contact: Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.153:5064 Jan 27 13:22:01 DEBUG[17464]: ##### Testing 204.8.226.3 with 0.0.0.0 Jan 27 13:22:01 DEBUG[17464]: ##### Testing 204.8.226.3 with 0.0.0.0 Jan 27 13:22:01 DEBUG[17464]: Auto destroying call '8b8930d4-bc517ce6-b185fa4f@199.104.120.153' Jan 27 13:22:01 VERBOSE[17464]: Destroying call '8b8930d4-bc517ce6-b185fa4f@199.104.120.153' Jan 27 13:22:03 DEBUG[17464]: Auto destroying call '4b4b94bc-23a93f8e-ba61b057@199.104.120.153' Jan 27 13:22:03 VERBOSE[17464]: Destroying call '4b4b94bc-23a93f8e-ba61b057@199.104.120.153' Jan 27 13:22:04 DEBUG[17464]: Manager received command 'redirect' Jan 27 13:22:04 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:22:04 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:22:04 VERBOSE[17464]: We're at 199.104.120.160 port 19974 Jan 27 13:22:04 VERBOSE[17464]: Answering/Requesting with root capability 0x4 (ulaw) Jan 27 13:22:04 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:22:04 VERBOSE[17464]: 11 headers, 10 lines Jan 27 13:22:04 VERBOSE[17464]: Reliably Transmitting: INVITE sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK20ea8106 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 104 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 222 v=0 o=root 17464 17466 IN IP4 199.104.120.160 s=session c=IN IP4 199.104.120.160 t=0 0 m=audio 19974 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 199.104.120.151:5060 Jan 27 13:22:04 DEBUG[17464]: Returning from native bridge, channels: SIP/139-8369, SIP/104-5a59 Jan 27 13:22:04 DEBUG[17464]: update_user_counter(104) - decrement outUse counter Jan 27 13:22:04 DEBUG[17464]: Exiting with DIALSTATUS=ANSWER. Jan 27 13:22:04 VERBOSE[17464]: == Spawn extension (office, 8669, 0) exited non-zero on 'SIP/139-8369' in macro 'stdexten' Jan 27 13:22:04 VERBOSE[17464]: == Spawn extension (office, 8669, 0) exited non-zero on 'SIP/139-8369' Jan 27 13:22:04 VERBOSE[17464]: -- Executing Playback("SIP/139-8369", "lyrics-louie-louie") in new stack Jan 27 13:22:04 VERBOSE[17464]: > Build translator: source=2/1(gsm); dest=4/2(ulaw) Jan 27 13:22:04 VERBOSE[17464]: > translator chain ->gsmtolin Jan 27 13:22:04 VERBOSE[17464]: > translator chain ->lintoulaw Jan 27 13:22:04 DEBUG[17464]: Scheduling timer at 160 sample intervals Jan 27 13:22:04 VERBOSE[17464]: -- Playing 'lyrics-louie-louie' (language 'en') Jan 27 13:22:04 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK20ea8106 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 CSeq: 104 INVITE Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Type: application/sdp Content-Length: 193 v=0 o=- 1106855049 1106855049 IN IP4 199.104.120.151 s=Polycom IP Phone c=IN IP4 199.104.120.151 t=0 0 m=audio 2222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:22:04 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:22:04 DEBUG[17464]: Acked pending invite 104 Jan 27 13:22:04 DEBUG[17464]: Stopping retransmission on '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' of Request 104: Found Jan 27 13:22:04 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:22:04 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:22:04 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.151:2222 Jan 27 13:22:04 DEBUG[17464]: Peer audio RTP is at port 199.104.120.151:2222 Jan 27 13:22:04 VERBOSE[17464]: Found description format PCMU Jan 27 13:22:04 VERBOSE[17464]: Found description format telephone-event Jan 27 13:22:04 VERBOSE[17464]: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:22:04 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:22:04 DEBUG[17464]: build_route: Retaining previous route: Jan 27 13:22:04 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:22:04 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:22:04 VERBOSE[17464]: Transmitting: ACK sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK02dd2059 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:22:04 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:22:04 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:22:04 VERBOSE[17464]: Reliably Transmitting: BYE sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK7af23802 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 Contact: Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 CSeq: 105 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:22:04 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK7af23802 From: "kevin" ;tag=as40143110 To: ;tag=71D35509-4C577D10 CSeq: 105 BYE Call-ID: 03840a242668880b2fcb9aa57329b4b9@199.104.120.160 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Length: 0 Jan 27 13:22:04 VERBOSE[17464]: 9 headers, 0 lines Jan 27 13:22:04 DEBUG[17464]: Stopping retransmission on '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' of Request 105: Found Jan 27 13:22:04 VERBOSE[17464]: Destroying call '03840a242668880b2fcb9aa57329b4b9@199.104.120.160' Jan 27 13:22:28 DEBUG[17464]: ##### Testing 209.63.147.87 with 0.0.0.0 Jan 27 13:22:28 DEBUG[17464]: ##### Testing 209.63.147.87 with 0.0.0.0 Jan 27 13:22:34 DEBUG[17464]: Immediately destroying 5, having received INVAL Jan 27 13:22:50 DEBUG[17464]: Scheduling timer at 0 sample intervals Jan 27 13:22:50 DEBUG[17464]: Scheduling timer at 0 sample intervals Jan 27 13:22:50 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:22:50 VERBOSE[17464]: -- Executing Hangup("SIP/139-8369", "") in new stack Jan 27 13:22:50 VERBOSE[17464]: == Spawn extension (office, 8669, 2) exited non-zero on 'SIP/139-8369' Jan 27 13:22:50 DEBUG[17464]: cdr_mysql: inserting a CDR record. Jan 27 13:22:50 DEBUG[17464]: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) VALUES ('2005-01-27 13:21:55','\"kevin\" <139>','139','8669','office', 'SIP/139-8369','SIP/104-5a59','Hangup','',55,53,'ANSWERED',3,'139','') Jan 27 13:22:50 DEBUG[17464]: update_user_counter(139) - decrement outUse counter Jan 27 13:22:50 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:22:50 VERBOSE[17464]: set_destination: set destination to 199.104.120.153, port 5064 Jan 27 13:22:50 VERBOSE[17464]: Reliably Transmitting: BYE sip:139@199.104.120.153:5064 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK707d7ca7;rport From: ;tag=as11d583bc To: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA Contact: Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.153:5064 Jan 27 13:22:50 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK707d7ca7;rport From: ;tag=as11d583bc To: "Kevin Blackham" ;tag=10D5A8DD-3513CFAA CSeq: 103 BYE Call-ID: b48d1759-85b9503-3000e7b8@199.104.120.153 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.3.1 Content-Length: 0 Jan 27 13:22:50 VERBOSE[17464]: 9 headers, 0 lines Jan 27 13:22:50 DEBUG[17464]: Stopping retransmission on 'b48d1759-85b9503-3000e7b8@199.104.120.153' of Request 103: Found Jan 27 13:22:50 VERBOSE[17464]: Destroying call 'b48d1759-85b9503-3000e7b8@199.104.120.153'