Jan 27 13:33:32 VERBOSE[17464]: Sip read: INVITE sip:104@pbx.xmission.com SIP/2.0 Via: SIP/2.0/UDP 199.104.120.154:5060;branch=z9hG4bK1dd399cb From: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a To: Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 253 Accept: application/sdp Remote-Party-ID: "receptionist" ;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 11536 26587 IN IP4 199.104.120.154 s=SIP Call c=IN IP4 199.104.120.154 t=0 0 m=audio 31750 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 0-15 Jan 27 13:33:32 VERBOSE[17464]: 13 headers, 11 lines Jan 27 13:33:32 VERBOSE[17464]: Using latest request as basis request Jan 27 13:33:32 VERBOSE[17464]: Sending to 199.104.120.154 : 5060 (non-NAT) Jan 27 13:33:32 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:33:32 VERBOSE[17464]: Found RTP audio format 8 Jan 27 13:33:32 VERBOSE[17464]: Found RTP audio format 18 Jan 27 13:33:32 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:33:32 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.154:31750 Jan 27 13:33:32 DEBUG[17464]: Peer audio RTP is at port 199.104.120.154:31750 Jan 27 13:33:32 VERBOSE[17464]: Found description format PCMU Jan 27 13:33:32 VERBOSE[17464]: Found description format PCMA Jan 27 13:33:32 VERBOSE[17464]: Found description format G729 Jan 27 13:33:32 VERBOSE[17464]: Found description format telephone-event Jan 27 13:33:32 VERBOSE[17464]: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:33:32 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:33:32 DEBUG[17464]: Setting NAT on RTP to 0 Jan 27 13:33:32 VERBOSE[17464]: Found user '100' Jan 27 13:33:32 DEBUG[17464]: Check for res for 100 Jan 27 13:33:32 DEBUG[17464]: Call from user '100' is 1 out of 0 Jan 27 13:33:32 VERBOSE[17464]: Looking for 104 in office Jan 27 13:33:32 DEBUG[17464]: build_route: Contact hop: Jan 27 13:33:32 VERBOSE[17464]: list_route: hop: Jan 27 13:33:32 VERBOSE[17464]: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 199.104.120.154:5060;branch=z9hG4bK1dd399cb From: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a To: ;tag=as6d057f57 Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 199.104.120.154:5060 Jan 27 13:33:32 VERBOSE[17464]: -- Executing Macro("SIP/100-6d96", "stdexten|104|hangup|pbx") in new stack Jan 27 13:33:32 VERBOSE[17464]: -- Executing DBget("SIP/100-6d96", "chan=EXT/104") in new stack Jan 27 13:33:32 VERBOSE[17464]: -- DBget: varname=chan, family=EXT, key=104 Jan 27 13:33:32 VERBOSE[17464]: -- DBget: set variable chan to SIP/104 Jan 27 13:33:32 VERBOSE[17464]: -- Executing GotoIf("SIP/100-6d96", "?6") in new stack Jan 27 13:33:32 DEBUG[17464]: Not taking any branch Jan 27 13:33:32 VERBOSE[17464]: -- Executing DBget("SIP/100-6d96", "cid_name=EXT/100/cid_name") in new stack Jan 27 13:33:32 VERBOSE[17464]: -- DBget: varname=cid_name, family=EXT, key=100/cid_name Jan 27 13:33:32 VERBOSE[17464]: -- DBget: set variable cid_name to ssant Jan 27 13:33:32 VERBOSE[17464]: -- Executing SetCIDName("SIP/100-6d96", "ssant") in new stack Jan 27 13:33:32 VERBOSE[17464]: -- Executing SetVar("SIP/100-6d96", "_ALERT_INFO=none") in new stack Jan 27 13:33:32 VERBOSE[17464]: -- Executing Dial("SIP/100-6d96", "SIP/104|25") in new stack Jan 27 13:33:32 DEBUG[17464]: Setting NAT on RTP to 0 Jan 27 13:33:32 DEBUG[17464]: Outgoing Call for 104 Jan 27 13:33:32 DEBUG[17464]: Call from user '104' is 1 out of 0 Jan 27 13:33:32 VERBOSE[17464]: We're at 199.104.120.160 port 10466 Jan 27 13:33:32 VERBOSE[17464]: Answering/Requesting with root capability 0x4 (ulaw) Jan 27 13:33:32 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:33:32 VERBOSE[17464]: 13 headers, 10 lines Jan 27 13:33:32 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=z9hG4bK09539056 From: "ssant" ;tag=as12a23dcc To: Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 27 Jan 2005 20:33:32 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 10466 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 13:33:32 VERBOSE[17464]: -- Called 104 Jan 27 13:33:32 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:32 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:32 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:32 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:32 VERBOSE[17464]: Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK09539056 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE CSeq: 102 INVITE Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Length: 0 Jan 27 13:33:32 VERBOSE[17464]: 9 headers, 0 lines Jan 27 13:33:32 DEBUG[17464]: (Provisional) Stopping retransmission (but retaining packet) on '2c853db414a3edba061b00ed42762d81@199.104.120.160' Request 102: Found Jan 27 13:33:32 VERBOSE[17464]: Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK09539056 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE CSeq: 102 INVITE Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Allow-Events: talk,hold,conference Content-Length: 0 Jan 27 13:33:32 VERBOSE[17464]: 10 headers, 0 lines Jan 27 13:33:32 DEBUG[17464]: (Provisional) Stopping retransmission (but retaining packet) on '2c853db414a3edba061b00ed42762d81@199.104.120.160' Request 102: Found Jan 27 13:33:32 VERBOSE[17464]: -- SIP/104-d890 is ringing Jan 27 13:33:32 VERBOSE[17464]: Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP 199.104.120.154:5060;branch=z9hG4bK1dd399cb From: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a To: ;tag=as6d057f57 Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 199.104.120.154:5060 Jan 27 13:33:33 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK09539056 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE CSeq: 102 INVITE Call-ID: 2c853db414a3edba061b00ed42762d81@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=- 1106855744 1106855744 IN IP4 199.104.120.151 s=Polycom IP Phone c=IN IP4 199.104.120.151 t=0 0 m=audio 2226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:33:33 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:33:33 DEBUG[17464]: Acked pending invite 102 Jan 27 13:33:33 DEBUG[17464]: Stopping retransmission on '2c853db414a3edba061b00ed42762d81@199.104.120.160' of Request 102: Found Jan 27 13:33:33 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:33:33 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:33:33 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.151:2226 Jan 27 13:33:33 DEBUG[17464]: Peer audio RTP is at port 199.104.120.151:2226 Jan 27 13:33:33 VERBOSE[17464]: Found description format PCMU Jan 27 13:33:33 VERBOSE[17464]: Found description format telephone-event Jan 27 13:33:33 VERBOSE[17464]: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:33:33 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:33:33 DEBUG[17464]: build_route: Contact hop: Jan 27 13:33:33 VERBOSE[17464]: list_route: hop: Jan 27 13:33:33 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:33 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:33:33 VERBOSE[17464]: Transmitting: ACK sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK12940c5f From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:33:33 VERBOSE[17464]: -- SIP/104-d890 answered SIP/100-6d96 Jan 27 13:33:33 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:33 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:33 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:33 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:33:33 VERBOSE[17464]: We're at 199.104.120.160 port 17074 Jan 27 13:33:33 VERBOSE[17464]: Answering with preferred capability 0x4 (ulaw) Jan 27 13:33:33 VERBOSE[17464]: Answering with preferred capability 0x2 (gsm) Jan 27 13:33:33 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:33:33 VERBOSE[17464]: Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.154:5060;branch=z9hG4bK1dd399cb From: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a To: ;tag=as6d057f57 Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 101 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 17074 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 199.104.120.154:5060 Jan 27 13:33:33 VERBOSE[17464]: -- Attempting native bridge of SIP/100-6d96 and SIP/104-d890 Jan 27 13:33:33 DEBUG[17464]: Deferring reinvite on '00127f73-f1ea0008-27598232-22edd796@199.104.120.154' Jan 27 13:33:33 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:33 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:33:33 VERBOSE[17464]: We're at 199.104.120.160 port 10466 Jan 27 13:33:33 VERBOSE[17464]: Answering/Requesting with root capability 0x4 (ulaw) Jan 27 13:33:33 VERBOSE[17464]: Answering with capability 0x8 (alaw) Jan 27 13:33:33 VERBOSE[17464]: Answering with capability 0x100 (g729) Jan 27 13:33:33 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:33:33 VERBOSE[17464]: 11 headers, 12 lines Jan 27 13:33:33 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=z9hG4bK1b347709 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@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 199.104.120.154 s=session c=IN IP4 199.104.120.154 t=0 0 m=audio 31750 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 13:33:33 VERBOSE[17464]: Sip read: ACK sip:104@199.104.120.160:5061 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.154:5060;branch=z9hG4bK20af1fd8 From: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a To: ;tag=as6d057f57 Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 101 ACK User-Agent: CSCO/7 Content-Length: 0 Jan 27 13:33:33 VERBOSE[17464]: 8 headers, 0 lines Jan 27 13:33:33 DEBUG[17464]: Stopping retransmission on '00127f73-f1ea0008-27598232-22edd796@199.104.120.154' of Response 101: Found Jan 27 13:33:33 DEBUG[17464]: Sending pending reinvite on '00127f73-f1ea0008-27598232-22edd796@199.104.120.154' Jan 27 13:33:33 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:33 VERBOSE[17464]: set_destination: set destination to 199.104.120.154, port 5060 Jan 27 13:33:33 VERBOSE[17464]: We're at 199.104.120.160 port 17074 Jan 27 13:33:33 VERBOSE[17464]: Answering with preferred capability 0x4 (ulaw) Jan 27 13:33:33 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:33:33 VERBOSE[17464]: 11 headers, 10 lines Jan 27 13:33:33 VERBOSE[17464]: Reliably Transmitting: INVITE sip:100@199.104.120.154:5060 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK52d741ab;rport From: ;tag=as6d057f57 To: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a Contact: Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 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 2226 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 199.104.120.154:5060 Jan 27 13:33:34 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK1b347709 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE CSeq: 103 INVITE Call-ID: 2c853db414a3edba061b00ed42762d81@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=- 1106855745 1106855745 IN IP4 199.104.120.151 s=Polycom IP Phone c=IN IP4 199.104.120.151 t=0 0 m=audio 2226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:33:34 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:33:34 DEBUG[17464]: Acked pending invite 103 Jan 27 13:33:34 DEBUG[17464]: Stopping retransmission on '2c853db414a3edba061b00ed42762d81@199.104.120.160' of Request 103: Found Jan 27 13:33:34 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:33:34 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:33:34 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.151:2226 Jan 27 13:33:34 DEBUG[17464]: Peer audio RTP is at port 199.104.120.151:2226 Jan 27 13:33:34 VERBOSE[17464]: Found description format PCMU Jan 27 13:33:34 VERBOSE[17464]: Found description format telephone-event Jan 27 13:33:34 VERBOSE[17464]: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:33:34 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:33:34 DEBUG[17464]: build_route: Retaining previous route: Jan 27 13:33:34 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:34 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:33:34 VERBOSE[17464]: Transmitting: ACK sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK6d5d5945 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:33:34 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK52d741ab;rport From: ;tag=as6d057f57 To: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 102 INVITE Server: CSCO/7 Contact: Content-Type: application/sdp Content-Length: 200 v=0 o=Cisco-SIPUA 78 18017 IN IP4 199.104.120.154 s=SIP Call c=IN IP4 199.104.120.154 t=0 0 m=audio 31750 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Jan 27 13:33:34 VERBOSE[17464]: 10 headers, 9 lines Jan 27 13:33:34 DEBUG[17464]: Acked pending invite 102 Jan 27 13:33:34 DEBUG[17464]: Stopping retransmission on '00127f73-f1ea0008-27598232-22edd796@199.104.120.154' of Request 102: Found Jan 27 13:33:34 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:33:34 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:33:34 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.154:31750 Jan 27 13:33:34 DEBUG[17464]: Peer audio RTP is at port 199.104.120.154:31750 Jan 27 13:33:34 VERBOSE[17464]: Found description format PCMU Jan 27 13:33:34 VERBOSE[17464]: Found description format telephone-event Jan 27 13:33:34 VERBOSE[17464]: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:33:34 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:33:34 DEBUG[17464]: build_route: Contact hop: Jan 27 13:33:34 VERBOSE[17464]: list_route: hop: Jan 27 13:33:34 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:34 VERBOSE[17464]: set_destination: set destination to 199.104.120.154, port 5060 Jan 27 13:33:34 VERBOSE[17464]: Transmitting: ACK sip:100@199.104.120.154:5060 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK2f853e43;rport From: ;tag=as6d057f57 To: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a Contact: Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.154:5060 Jan 27 13:33:39 NOTICE[17464]: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 Jan 27 13:33:41 DEBUG[17464]: ##### Testing 204.8.226.3 with 0.0.0.0 Jan 27 13:33:41 DEBUG[17464]: ##### Testing 204.8.226.3 with 0.0.0.0 Jan 27 13:33:44 VERBOSE[17464]: -- Registered to '69.73.19.178', who sees us as 199.104.120.160:4569 Jan 27 13:33:45 DEBUG[17464]: Manager received command 'redirect' Jan 27 13:33:45 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:45 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:33:45 VERBOSE[17464]: We're at 199.104.120.160 port 10466 Jan 27 13:33:45 VERBOSE[17464]: Answering/Requesting with root capability 0x4 (ulaw) Jan 27 13:33:45 VERBOSE[17464]: Answering with non-codec capability 0x1 (telephone-event) Jan 27 13:33:45 VERBOSE[17464]: 11 headers, 10 lines Jan 27 13:33:45 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=z9hG4bK097a7ca8 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@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 10466 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 13:33:45 DEBUG[17464]: Returning from native bridge, channels: SIP/100-6d96, SIP/104-d890 Jan 27 13:33:45 DEBUG[17464]: update_user_counter(104) - decrement outUse counter Jan 27 13:33:45 DEBUG[17464]: Exiting with DIALSTATUS=ANSWER. Jan 27 13:33:45 VERBOSE[17464]: == Spawn extension (office, 8669, 0) exited non-zero on 'SIP/100-6d96' in macro 'stdexten' Jan 27 13:33:45 VERBOSE[17464]: == Spawn extension (office, 8669, 0) exited non-zero on 'SIP/100-6d96' Jan 27 13:33:45 VERBOSE[17464]: -- Executing Playback("SIP/100-6d96", "lyrics-louie-louie") in new stack Jan 27 13:33:45 VERBOSE[17464]: > Build translator: source=2/1(gsm); dest=4/2(ulaw) Jan 27 13:33:45 VERBOSE[17464]: > translator chain ->gsmtolin Jan 27 13:33:45 VERBOSE[17464]: > translator chain ->lintoulaw Jan 27 13:33:45 DEBUG[17464]: Scheduling timer at 160 sample intervals Jan 27 13:33:45 VERBOSE[17464]: -- Playing 'lyrics-louie-louie' (language 'en') Jan 27 13:33:45 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK097a7ca8 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE CSeq: 104 INVITE Call-ID: 2c853db414a3edba061b00ed42762d81@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=- 1106855746 1106855746 IN IP4 199.104.120.151 s=Polycom IP Phone c=IN IP4 199.104.120.151 t=0 0 m=audio 2226 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 Jan 27 13:33:45 VERBOSE[17464]: 11 headers, 8 lines Jan 27 13:33:45 DEBUG[17464]: Acked pending invite 104 Jan 27 13:33:45 DEBUG[17464]: Stopping retransmission on '2c853db414a3edba061b00ed42762d81@199.104.120.160' of Request 104: Found Jan 27 13:33:45 VERBOSE[17464]: Found RTP audio format 0 Jan 27 13:33:45 VERBOSE[17464]: Found RTP audio format 101 Jan 27 13:33:45 VERBOSE[17464]: Peer audio RTP is at port 199.104.120.151:2226 Jan 27 13:33:45 DEBUG[17464]: Peer audio RTP is at port 199.104.120.151:2226 Jan 27 13:33:45 VERBOSE[17464]: Found description format PCMU Jan 27 13:33:45 VERBOSE[17464]: Found description format telephone-event Jan 27 13:33:45 VERBOSE[17464]: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Jan 27 13:33:45 VERBOSE[17464]: Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Jan 27 13:33:45 DEBUG[17464]: build_route: Retaining previous route: Jan 27 13:33:45 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:45 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:33:45 VERBOSE[17464]: Transmitting: ACK sip:104@199.104.120.151 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK7de03510 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:33:45 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:33:45 VERBOSE[17464]: set_destination: set destination to 199.104.120.151, port 5060 Jan 27 13:33:45 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=z9hG4bK02572456 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE Contact: Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 CSeq: 105 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.151:5060 Jan 27 13:33:46 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK02572456 From: "ssant" ;tag=as12a23dcc To: ;tag=D3151367-77F6E0EE CSeq: 105 BYE Call-ID: 2c853db414a3edba061b00ed42762d81@199.104.120.160 Contact: User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Content-Length: 0 Jan 27 13:33:46 VERBOSE[17464]: 9 headers, 0 lines Jan 27 13:33:46 DEBUG[17464]: Stopping retransmission on '2c853db414a3edba061b00ed42762d81@199.104.120.160' of Request 105: Found Jan 27 13:33:46 VERBOSE[17464]: Destroying call '2c853db414a3edba061b00ed42762d81@199.104.120.160' Jan 27 13:34:08 DEBUG[17464]: ##### Testing 209.63.147.87 with 0.0.0.0 Jan 27 13:34:08 DEBUG[17464]: ##### Testing 209.63.147.87 with 0.0.0.0 Jan 27 13:34:31 DEBUG[17464]: ##### Testing 204.8.226.3 with 0.0.0.0 Jan 27 13:34:31 DEBUG[17464]: ##### Testing 204.8.226.3 with 0.0.0.0 Jan 27 13:34:31 DEBUG[17464]: Scheduling timer at 0 sample intervals Jan 27 13:34:31 DEBUG[17464]: Scheduling timer at 0 sample intervals Jan 27 13:34:31 VERBOSE[17464]: > Build translator: source=4/2(ulaw); dest=4/2(ulaw) Jan 27 13:34:31 VERBOSE[17464]: -- Executing Hangup("SIP/100-6d96", "") in new stack Jan 27 13:34:31 VERBOSE[17464]: == Spawn extension (office, 8669, 2) exited non-zero on 'SIP/100-6d96' Jan 27 13:34:31 DEBUG[17464]: cdr_mysql: inserting a CDR record. Jan 27 13:34:31 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 13:33:32','\"ssant\" <100>','100','8669','office', 'SIP/100-6d96','SIP/104-d890','Hangup','',59,58,'ANSWERED',3,'100','') Jan 27 13:34:31 DEBUG[17464]: update_user_counter(100) - decrement outUse counter Jan 27 13:34:31 VERBOSE[17464]: set_destination: Parsing for address/port to send to Jan 27 13:34:31 VERBOSE[17464]: set_destination: set destination to 199.104.120.154, port 5060 Jan 27 13:34:31 VERBOSE[17464]: Reliably Transmitting: BYE sip:100@199.104.120.154:5060 SIP/2.0 Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK41f740ba;rport From: ;tag=as6d057f57 To: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a Contact: Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 199.104.120.154:5060 Jan 27 13:34:31 VERBOSE[17464]: Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 199.104.120.160:5061;branch=z9hG4bK41f740ba;rport From: ;tag=as6d057f57 To: "receptionist" ;tag=00127f73f1ea00021d3d1580-26305a5a Call-ID: 00127f73-f1ea0008-27598232-22edd796@199.104.120.154 CSeq: 103 BYE Server: CSCO/7 Content-Length: 0 RTP-RxStat: Dur=58,Pkt=2890,Oct=462400,LatePkt=0,LostPkt=294158,AvgJit=0 RTP-TxStat: Dur=58,Pkt=2898,Oct=463680 Jan 27 13:34:31 VERBOSE[17464]: 10 headers, 0 lines Jan 27 13:34:31 DEBUG[17464]: Stopping retransmission on '00127f73-f1ea0008-27598232-22edd796@199.104.120.154' of Request 103: Found Jan 27 13:34:31 VERBOSE[17464]: Destroying call '00127f73-f1ea0008-27598232-22edd796@199.104.120.154'