SIP Debugging Enabled for IP: 209.114.219.98:5060 asterisk*CLI> Sip read: INVITE sip:1001@209.114.219.69;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.169:5060 From: 3213084003 ;tag=2570885485 To: Call-ID: 2724590751@192.168.200.169 CSeq: 1 INVITE Contact: 3213084003 User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Expires: 300 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 282 Content-Type: application/sdp v=0 o=3213084003 36084 36084 IN IP4 192.168.200.169 s=ATA186 Call c=IN IP4 192.168.200.169 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 12 lines Using latest request as basis request Found user '3213084003' Looking for 1001 in default list_route: hop: Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060 From: 3213084003 ;tag=2570885485 To: ;tag=as5456d442 Call-ID: 2724590751@192.168.200.169 CSeq: 1 INVITE User-Agent: HCC Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 209.114.219.98:5060 -- Executing MYSQL("SIP/3213084003-f6ba", "Connect connid localhost asterisk longpoint asterisk") in new stack -- Executing MYSQL("SIP/3213084003-f6ba", "Query resultid 117 SELECT scriptname from mac2pin where userid=3213084003") in new stack -- Executing MYSQL("SIP/3213084003-f6ba", "Fetch fetchid 118 AGIScript") in new stack Jan 5 20:20:55 WARNING[3032]: app_addon_sql_mysql.c:318 aMYSQL_fetch: ast_MYSQL_fetch: numFields=1 -- Executing GotoIf("SIP/3213084003-f6ba", "0?5:7") in new stack -- Goto (default,1001,7) -- Executing AGI("SIP/3213084003-f6ba", "HCC_TEST.agi|1001") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/HCC_TEST.agi HCC_TEST.agi|1001: Dialing cisco SIP for Mark -- AGI Script Executing Application: (Dial) Options: (SIP/003214093773@3213084999) -- Called 003214093773@3213084999 -- SIP/3213084999-ee69 is making progress passing it to SIP/3213084003-f6ba We're at 209.114.219.69 port 17550 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Transmitting (NAT): SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060 From: 3213084003 ;tag=2570885485 To: ;tag=as5456d44 Call-ID: 2724590751@192.168.200.169 CSeq: 1 INVITE User-Agent: HCC Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=root 3032 3032 IN IP4 209.114.219.69 s=session c=IN IP4 209.114.219.69 t=0 0 m=audio 17550 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 209.114.219.98:5060 -- SIP/3213084999-ee69 answered SIP/3213084003-f6ba We're at 209.114.219.69 port 17550 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060 From: 3213084003 ;tag=2570885485 To: ;tag=as5456d442 Call-ID: 2724590751@192.168.200.169 CSeq: 1 INVITE User-Agent: HCC Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=root 3032 3033 IN IP4 209.114.219.69 s=session c=IN IP4 209.114.219.69 t=0 0 m=audio 17550 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 209.114.219.98:5060 -- Attempting native bridge of SIP/3213084003-f6ba and SIP/3213084999-ee69 asterisk*CLI> Sip read: ACK sip:1001@209.114.219.69 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.169:5060 From: 3213084003 ;tag=2570885485 To: ;tag=as5456d442 Call-ID: 2724590751@192.168.200.169 CSeq: 1 ACK User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 8 headers, 0 lines set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.169, port 5060 We're at 209.114.219.69 port 17550 Answering with preferred capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 10 lines Reliably Transmitting: INVITE sip:3213084003@192.168.200.169:5060 SIP/2.0 Via: SIP/2.0/UDP 209.114.219.69:5060;branch=z9hG4bK6c2695c0;rport From: ;tag=as5456d442 To: 3213084003 ;tag=2570885485 Contact: Call-ID: 2724590751@192.168.200.169 CSeq: 102 INVITE User-Agent: HCC Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 v=0 o=root 3032 3034 IN IP4 209.114.219.90 s=session c=IN IP4 209.114.219.90 t=0 0 m=audio 19766 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (NAT) to 209.114.219.98:5060 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.114.219.69:5060;branch=z9hG4bK6c2695c0;rport From: ;tag=as5456d442 To: 3213084003 ;tag=2570885485 Call-ID: 2724590751@192.168.200.169 CSeq: 102 INVITE Contact: 3213084003 Server: Cisco ATA 186 v3.1.0 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 230 Content-Type: application/sdp v=0 o=3213084003 36133 36133 IN IP4 192.168.200.169 s=ATA186 Call c=IN IP4 192.168.200.169 t=0 0 m=audio 16384 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 10 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.200.169:16384 Found description format G729 Found description format telephone-event Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.200.169, port 5060 Transmitting: ACK sip:3213084003@192.168.200.169:5060 SIP/2.0 Via: SIP/2.0/UDP 209.114.219.69:5060;branch=z9hG4bK28262899;rport From: ;tag=as5456d442 To: 3213084003 ;tag=2570885485 Contact: Call-ID: 2724590751@192.168.200.169 CSeq: 102 ACK User-Agent: HCC Asterisk PBX Content-Length: 0 (NAT) to 209.114.219.98:5060 asterisk*CLI> Sip read: BYE sip:1001@209.114.219.69 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.169:5060 From: 3213084003 ;tag=2570885485 To: ;tag=as5456d442 Call-ID: 2724590751@192.168.200.169 CSeq: 2 BYE User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 8 headers, 0 lines Sending to 192.168.200.169 : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060 From: 3213084003 ;tag=2570885485 To: ;tag=as5456d442 Call-ID: 2724590751@192.168.200.169 CSeq: 2 BYE User-Agent: HCC Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 209.114.219.98:5060 -- AGI Script HCC_TEST.agi completed, returning 0 Destroying call '2724590751@192.168.200.169' asterisk*CLI>