Asterisk*CLI> <-- SIP read from 200.75.231.230:31878: INVITE sip:1018@200.75.200.134 SIP/2.0 Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-831153065f4e243b-1--d87543-;rport Max-Forwards: 70 Contact: To: "Jorge Oscar Pinzón" From: "Irving Barría";tag=b222d708 Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 295 v=0 o=- 7 2 IN IP4 200.75.231.230 s= c=IN IP4 200.75.231.230 t=0 0 m=audio 32800 RTP/AVP 3 101 a=alt:1 1 : Bpj08+r1 wc1vyVzh 192.168.129.163 2198 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:29AA36266B324C23B2591F7320DCEE43 --- (12 headers 11 lines)--- Using INVITE request as basis request - 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. Sending to 200.75.231.230 : 31878 (NAT) Found peer '1000' Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 200.75.231.230:32800 Found description format telephone-event Capabilities: us - 0x80002 (gsm|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 1018 in from-sip (domain 200.75.200.134) list_route: hop: Transmitting (NAT) to 200.75.231.230:31878: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-831153065f4e243b-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=b222d708 To: "Jorge Oscar Pinzón" Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 1 INVITE User-Agent: Nimbus SIP Server v2.00.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/1000-b33d", "callprocessing|1018|SIP/1018") in new stack -- Executing Set("SIP/1000-b33d", "LANGUAGE()=es") in new stack -- Executing Set("SIP/1000-b33d", "DB(LastCalledExt/1000)=1018") in new stack -- Executing Set("SIP/1000-b33d", "DB(LastCallingExt/1018)=1000") in new stack -- Executing Dial("SIP/1000-b33d", "SIP/1018|30") in new stack We're at 200.75.200.134 port 15168 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 10 lines Reliably Transmitting (NAT) to 201.221.243.70:34000: INVITE sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK2548f4b4 From: "Irving Barría" ;tag=as71d339d2 To: Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 102 INVITE User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Date: Wed, 12 Jul 2006 17:00:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 219 v=0 o=root 20147 20147 IN IP4 200.75.200.134 s=session c=IN IP4 200.75.200.134 t=0 0 m=audio 15168 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 1018 Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK2548f4b4 Contact: To: ;tag=b01e796f From: "Irving Barría";tag=as71d339d2 Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 102 INVITE User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/1018-1eb1 is ringing Transmitting (NAT) to 200.75.231.230:31878: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-831153065f4e243b-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=b222d708 To: "Jorge Oscar Pinzón";tag=as42e5095b Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 1 INVITE User-Agent: Nimbus SIP Server v2.00.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK2548f4b4 Contact: To: ;tag=b01e796f From: "Irving Barría";tag=as71d339d2 Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 243 v=0 o=- 1 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 53012 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:022156D09ADE4863B60652A4D5DE2E0D --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:53012 Found description format telephone-event Capabilities: us - 0x80002 (gsm|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 Transmitting (NAT) to 201.221.243.70:34000: ACK sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK06fff68b From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 102 ACK User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/1018-1eb1 answered SIP/1000-b33d We're at 200.75.200.134 port 15270 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 200.75.231.230:31878: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-831153065f4e243b-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=b222d708 To: "Jorge Oscar Pinzón";tag=as42e5095b Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 1 INVITE User-Agent: Nimbus SIP Server v2.00.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 219 v=0 o=root 20147 20147 IN IP4 200.75.200.134 s=session c=IN IP4 200.75.200.134 t=0 0 m=audio 15270 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1000-b33d and SIP/1018-1eb1 set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 We're at 200.75.200.134 port 15168 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 10 lines Reliably Transmitting (NAT) to 201.221.243.70:34000: INVITE sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK71ba0710 From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 103 INVITE User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 219 v=0 o=root 20147 20148 IN IP4 200.75.231.230 s=session c=IN IP4 200.75.231.230 t=0 0 m=audio 32800 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Asterisk*CLI> <-- SIP read from 200.75.231.230:31878: ACK sip:1018@200.75.200.134 SIP/2.0 Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-e2013a55e875f078-1--d87543-;rport Max-Forwards: 70 Contact: To: "Jorge Oscar Pinzón";tag=as42e5095b From: "Irving Barría";tag=b222d708 Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 1 ACK User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 0 --- (10 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 200.75.231.230, port 40778 We're at 200.75.200.134 port 15270 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (NAT) to 200.75.231.230:31878: INVITE sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK4efb3ec2 From: "Jorge Oscar Pinzón";tag=as42e5095b To: "Irving Barría";tag=b222d708 Contact: Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 102 INVITE User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 219 v=0 o=root 20147 20148 IN IP4 201.221.243.70 s=session c=IN IP4 201.221.243.70 t=0 0 m=audio 53012 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Asterisk*CLI> <-- SIP read from 200.75.231.230:31878: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK4efb3ec2 Contact: To: "Irving Barría";tag=b222d708 From: "Jorge Oscar Pinzón";tag=as42e5095b Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 243 v=0 o=- 7 3 IN IP4 200.75.231.230 s= c=IN IP4 200.75.231.230 t=0 0 m=audio 32800 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:29AA36266B324C23B2591F7320DCEE43 --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 200.75.231.230:32800 Found description format telephone-event Capabilities: us - 0x80002 (gsm|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 200.75.231.230, port 40778 Transmitting (NAT) to 200.75.231.230:31878: ACK sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK2180314d From: "Jorge Oscar Pinzón";tag=as42e5095b To: "Irving Barría";tag=b222d708 Contact: Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 102 ACK User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK2548f4b4 Contact: To: ;tag=b01e796f From: "Irving Barría";tag=as71d339d2 Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 243 v=0 o=- 1 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 53012 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:022156D09ADE4863B60652A4D5DE2E0D --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:53012 Found description format telephone-event Capabilities: us - 0x80002 (gsm|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 Transmitting (NAT) to 201.221.243.70:34000: ACK sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK6e8b6be8 From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 102 ACK User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Content-Length: 0 --- Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK71ba0710 Contact: To: ;tag=b01e796f From: "Irving Barría";tag=as71d339d2 Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 243 v=0 o=- 1 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 53012 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:022156D09ADE4863B60652A4D5DE2E0D --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:53012 Found description format telephone-event Capabilities: us - 0x80002 (gsm|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 Transmitting (NAT) to 201.221.243.70:34000: ACK sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK2535f5f5 From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 103 ACK User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Content-Length: 0 --- Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: --- (0 headers 1 lines)--- Asterisk*CLI> <-- SIP read from 200.75.231.230:31878: BYE sip:1018@200.75.200.134 SIP/2.0 Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-e05d085101016c77-1--d87543-;rport Max-Forwards: 70 Contact: To: "Jorge Oscar Pinzón";tag=as42e5095b From: "Irving Barría";tag=b222d708 Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 2 BYE User-Agent: X-Lite release 1002tx stamp 29712 Reason: SIP;description="User Hung Up" Content-Length: 0 --- (11 headers 0 lines)--- Sending to 200.75.231.230 : 31878 (NAT) Transmitting (NAT) to 200.75.231.230:31878: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.231.230:31878;branch=z9hG4bK-d87543-e05d085101016c77-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=b222d708 To: "Jorge Oscar Pinzón";tag=as42e5095b Call-ID: 05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 2 BYE User-Agent: Nimbus SIP Server v2.00.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 We're at 200.75.200.134 port 15168 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 10 lines Reliably Transmitting (NAT) to 201.221.243.70:34000: INVITE sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK5e692e87 From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 104 INVITE User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 219 v=0 o=root 20147 20149 IN IP4 200.75.200.134 s=session c=IN IP4 200.75.200.134 t=0 0 m=audio 15168 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- == Spawn extension (macro-callprocessing, s, 4) exited non-zero on 'SIP/1000-b33d' in macro 'callprocessing' == Spawn extension (macro-callprocessing, s, 4) exited non-zero on 'SIP/1000-b33d' --- Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK5e692e87 Contact: To: ;tag=b01e796f From: "Irving Barría";tag=as71d339d2 Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 243 v=0 o=- 1 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 53012 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:022156D09ADE4863B60652A4D5DE2E0D --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:53012 Found description format telephone-event Capabilities: us - 0x80002 (gsm|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 Transmitting (NAT) to 201.221.243.70:34000: ACK sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK456713a2 From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 104 ACK User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 201.221.243.70, port 34000 Reliably Transmitting (NAT) to 201.221.243.70:34000: BYE sip:1018@201.221.243.70:34000;rinstance=76c3162227302c47 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK27a1de8f From: "Irving Barría" ;tag=as71d339d2 To: ;tag=b01e796f Contact: Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 105 BYE User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Remote-Party-ID: "Irving Barría" ;privacy=off;screen=no Content-Length: 0 --- Destroying call '05635361937d993eNWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ.' --- (9 headers 0 lines)--- Asterisk*CLI> <-- SIP read from 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK27a1de8f Contact: To: ;tag=b01e796f From: "Irving Barría";tag=as71d339d2 Call-ID: 7397548f5806bb41741dbba36351cd18@200.75.200.134 CSeq: 105 BYE User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '7397548f5806bb41741dbba36351cd18@200.75.200.134'