pbx*CLI> <-- SIP read from 70.34.104.240:5060: INVITE sip:*0@pbx.contractcsi.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-526eeb72 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 101 INVITE Max-Forwards: 70 Contact: "Michael Azarian" Expires: 240 User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 6210936 6210936 IN IP4 192.168.0.248 s=- c=IN IP4 192.168.0.248 t=0 0 m=audio 16470 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (13 headers 18 lines)--- Using INVITE request as basis request - 1229c09d-f20167fe@192.168.0.248 Sending to 192.168.0.248 : 5060 (non-NAT) Reliably Transmitting (NAT) to 70.34.104.240:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-526eeb72;received=70.34.104.240 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as7456533d Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6871e45d" Content-Length: 0 --- Scheduling destruction of call '1229c09d-f20167fe@192.168.0.248' in 15000 ms Found user '30' pbx*CLI> <-- SIP read from 70.34.104.240:5060: ACK sip:*0@pbx.contractcsi.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-526eeb72 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as7456533d Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 101 ACK Max-Forwards: 70 Contact: "Michael Azarian" User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (10 headers 0 lines)--- pbx*CLI> <-- SIP read from 70.34.104.240:5060: INVITE sip:*0@pbx.contractcsi.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="30",realm="asterisk",nonce="6871e45d",uri="sip:*0@pbx.contractcsi.com",algorithm=MD5,response="ebcabeaeccb17ac09e4a86dd9f13f5ff" Contact: "Michael Azarian" Expires: 240 User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 399 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 6210936 6210936 IN IP4 192.168.0.248 s=- c=IN IP4 192.168.0.248 t=0 0 m=audio 16470 RTP/AVP 0 2 4 8 18 96 97 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 pbx*CLI> a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (14 headers 18 lines)--- Using INVITE request as basis request - 1229c09d-f20167fe@192.168.0.248 Sending to 192.168.0.248 : 5060 (NAT) Found user '30' Found RTP audio format 0 Found RTP audio format 2 pbx*CLI> Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.248:16470 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for *0 in internal-admin (domain pbx.contractcsi.com) list_route: hop: Transmitting (NAT) to 70.34.104.240:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e;received=70.34.104.240 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- pbx*CLI> -- Executing Set("SIP/30-093a52d0", "TIMEOUT(absolute) = 25") in new stack -- Executing AGI("SIP/30-093a52d0", "page.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/page.agi pbx*CLI> -- Remote UNIX connection pbx*CLI> -- Remote UNIX connection disconnected pbx*CLI> -- AGI Script page.agi completed, returning 0 -- Executing SetCallerID("SIP/30-093a52d0", ""PAGE: Michael Azarian" <") in new stack -- Executing Set("SIP/30-093a52d0", "_ALERT_INFO="Ring Answer"") in new stack -- Executing SIPAddHeader("SIP/30-093a52d0", "Call-Info: ;answer-after=0") in new stack -- Executing Page("SIP/30-093a52d0", "SIP/18&SIP/17&SIP/16&SIP/13&SIP/12&SIP/11") in new stack pbx*CLI> We're at 70.167.155.210 port 10132 pbx*CLI> Adding codec 0x4 (ulaw) to SDP pbx*CLI> Adding codec 0x8 (alaw) to SDP pbx*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx*CLI> Transmitting (NAT) to 70.34.104.240:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e;received=70.34.104.240 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as2456f08e Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2501 2501 IN IP4 70.167.155.210 s=session c=IN IP4 70.167.155.210 t=0 0 m=audio 10132 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 - - - - --- pbx*CLI> -- Playing 'beep' (language 'en') pbx*CLI> We're at 192.168.16.50 port 18620 pbx*CLI> Adding codec 0x4 (ulaw) to SDP pbx*CLI> Adding codec 0x8 (alaw) to SDP pbx*CLI> Adding codec 0x2 (gsm) to SDP pbx*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx*CLI> 15 headers, 12 lines pbx*CLI> Reliably Transmitting (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> We're at 192.168.16.50 port 13810 pbx*CLI> Adding codec 0x4 (ulaw) to SDP pbx*CLI> Adding codec 0x8 (alaw) to SDP pbx*CLI> Adding codec 0x2 (gsm) to SDP pbx*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx*CLI> 15 headers, 12 lines pbx*CLI> Reliably Transmitting (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> We're at 192.168.16.50 port 19550 pbx*CLI> Adding codec 0x4 (ulaw) to SDP pbx*CLI> Adding codec 0x8 (alaw) to SDP pbx*CLI> Adding codec 0x2 (gsm) to SDP pbx*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx*CLI> 15 headers, 12 lines pbx*CLI> Reliably Transmitting (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> We're at 192.168.16.50 port 16732 pbx*CLI> Adding codec 0x4 (ulaw) to SDP pbx*CLI> Adding codec 0x8 (alaw) to SDP pbx*CLI> Adding codec 0x2 (gsm) to SDP pbx*CLI> Adding non-codec 0x1 (telephone-event) to SDP pbx*CLI> 15 headers, 12 lines pbx*CLI> Reliably Transmitting (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> We're at 192.168.16.50 port 14124 pbx*CLI> Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP pbx*CLI> Adding non-codec 0x1 (telephone-event) to SDP 15 headers, 12 lines pbx*CLI> Reliably Transmitting (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- We're at 192.168.16.50 port 17244 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 15 headers, 12 lines pbx*CLI> Reliably Transmitting (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #1 (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #1 (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #2 (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #2 (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #2 (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #2 (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #3 (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #3 (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #4 (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #4 (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #4 (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #4 (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #5 (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #5 (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #5 (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #5 (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #5 (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #5 (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> We're at 70.167.155.210 port 10132 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 70.34.104.240:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-e963f67e;received=70.34.104.240 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as2456f08e Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 102 INVITE User-Agent: Asterisk PBX A pbx*CLI> llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 2501 2502 IN IP4 70.167.155.210 s=session c=IN IP4 70.167.155.210 t=0 0 m=audio 10132 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 - - - - --- -- Created MeetMe conference 1023 for conference '2093530406d' pbx*CLI> <-- SIP read from 70.34.104.240:5060: ACK sip:*0@70.167.155.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-b45543d3 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as2456f08e Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="30",realm="asterisk",nonce="6871e45d",uri="sip:*0@70.167.155.210",algorithm=MD5,response="1ba3e86baa691428ac0899e060178a87" Contact: "Michael Azarian" User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 0 --- (11 headers 0 lines)--- pbx*CLI> Retransmitting #6 (NAT) to 192.168.16.84:5060: INVITE sip:11@192.168.16.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2cf6ea14;rport From: "PAGE: Michael Azarian" <" ;tag=as5beed5a7 To: Contact: Call-ID: 4359e5a262f4e0773c88216517033e67@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 14124 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #6 (NAT) to 192.168.16.13:5060: INVITE sip:13@192.168.16.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK17859bb8;rport From: "PAGE: Michael Azarian" <" ;tag=as335b3254 To: Contact: Call-ID: 22caddf1659a1715126ea5a71869692e@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 19550 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #6 (no NAT) to 192.168.16.80:5060: INVITE sip:18@192.168.16.80:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK2437eb4d;rport From: "PAGE: Michael Azarian" <" ;tag=as20cf0eac To: Contact: Call-ID: 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 18620 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #6 (NAT) to 192.168.16.72:5060: INVITE sip:12@192.168.16.72:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK6f9cf33d;rport From: "PAGE: Michael Azarian" <" ;tag=as3a3f0c93 To: Contact: Call-ID: 226cd87216c32baf3e48049921efe777@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 16732 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #6 (NAT) to 192.168.16.16:5060: INVITE sip:16@192.168.16.16:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK4c80725c;rport From: "PAGE: Michael Azarian" <" ;tag=as06705b60 To: Contact: Call-ID: 0603149d18b7109627d0efe84987bbb5@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 17244 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Retransmitting #6 (no NAT) to 192.168.16.81:5060: INVITE sip:17@192.168.16.81:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.16.50:5060;branch=z9hG4bK59be9ca4;rport From: "PAGE: Michael Azarian" <" ;tag=as295fd97d To: Contact: Call-ID: 11bafa99125866ac60c082113ef7d971@192.168.16.50 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 12 Sep 2006 06:50:22 GMT Alert-Info: "Ring Answer" Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Call-Info: ;answer-after=0 Content-Type: application/sdp Content-Length: 263 v=0 o=root 2501 2501 IN IP4 192.168.16.50 s=session c=IN IP4 192.168.16.50 t=0 0 m=audio 13810 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pbx*CLI> Sep 11 23:50:23 WARNING[2541]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 4359e5a262f4e0773c88216517033e67@192.168.16.50 for seqno 102 (Critical Request) pbx*CLI> Sep 11 23:50:23 WARNING[2541]: chan_sip.c:1234 retrans_pkt: Hanging up call 4359e5a262f4e0773c88216517033e67@192.168.16.50 - no reply to our critical packet. pbx*CLI> Sep 11 23:50:23 WARNING[2541]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 22caddf1659a1715126ea5a71869692e@192.168.16.50 for seqno 102 (Critical Request) pbx*CLI> Sep 11 23:50:23 WARNING[2541]: chan_sip.c:1234 retrans_pkt: Hanging up call 22caddf1659a1715126ea5a71869692e@192.168.16.50 - no reply to our critical packet. pbx*CLI> Destroying call '4359e5a262f4e0773c88216517033e67@192.168.16.50' pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 for seqno 102 (Critical Request) pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1234 retrans_pkt: Hanging up call 5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50 - no reply to our critical packet. pbx*CLI> Destroying call '22caddf1659a1715126ea5a71869692e@192.168.16.50' pbx*CLI> Destroying call '5a7b1d02045dd0a55ec57f9f10ed2de6@192.168.16.50' pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 226cd87216c32baf3e48049921efe777@192.168.16.50 for seqno 102 (Critical Request) pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1234 retrans_pkt: Hanging up call 226cd87216c32baf3e48049921efe777@192.168.16.50 - no reply to our critical packet. pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 0603149d18b7109627d0efe84987bbb5@192.168.16.50 for seqno 102 (Critical Request) pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1234 retrans_pkt: Hanging up call 0603149d18b7109627d0efe84987bbb5@192.168.16.50 - no reply to our critical packet. pbx*CLI> Destroying call '0603149d18b7109627d0efe84987bbb5@192.168.16.50' pbx*CLI> Destroying call '226cd87216c32baf3e48049921efe777@192.168.16.50' pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 11bafa99125866ac60c082113ef7d971@192.168.16.50 for seqno 102 (Critical Request) pbx*CLI> Sep 11 23:50:24 WARNING[2541]: chan_sip.c:1234 retrans_pkt: Hanging up call 11bafa99125866ac60c082113ef7d971@192.168.16.50 - no reply to our critical packet. pbx*CLI> Destroying call '11bafa99125866ac60c082113ef7d971@192.168.16.50' pbx*CLI> <-- SIP read from 70.34.104.240:5060: BYE sip:*0@70.167.155.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-d2c5d1ac From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as2456f08e Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="30",realm="asterisk",nonce="6871e45d",uri="sip:*0@70.167.155.210",algorithm=MD5,response="e4ada15de33eaa86a331e23546f25827" User-Agent: Sipura/SPA942-4.1.10(e) Content-Length: 0 pbx*CLI> --- (10 headers 0 lines)--- pbx*CLI> Sending to 192.168.0.248 : 5060 (NAT) pbx*CLI> Transmitting (NAT) to 70.34.104.240:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.248:5060;branch=z9hG4bK-d2c5d1ac;received=70.34.104.240 From: "Michael Azarian" ;tag=103a80113b508f1ao0 To: ;tag=as2456f08e Call-ID: 1229c09d-f20167fe@192.168.0.248 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- pbx*CLI> -- Hungup 'Zap/pseudo-1544701011' pbx*CLI> == Spawn extension (internal-admin, *0, 6) exited non-zero on 'SIP/30-093a52d0' pbx*CLI> -- Executing Hangup("SIP/30-093a52d0", "") in new stack == Spawn extension (internal-admin, h, 1) exited non-zero on 'SIP/30-093a52d0' pbx*CLI> Destroying call '1229c09d-f20167fe@192.168.0.248' pbx*CLI> sip no debug pbx*CLI> SIP Debugging Disabled pbx*CLI>