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-d87543-9624df6f75622b0a-1--d87543-;rport Max-Forwards: 70 Contact: To: "Jorge Oscar Pinzón" From: "Irving Barría";tag=ee67d338 Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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: 296 v=0 o=- 0 2 IN IP4 200.75.231.230 s= c=IN IP4 200.75.231.230 t=0 0 m=audio 32778 RTP/AVP 3 101 a=alt:1 1 : gu0eDq7T Dy4o7+50 192.168.129.163 49986 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:1551168D3C8143148392210A0FE6342D --- (12 headers 11 lines)--- Using INVITE request as basis request - 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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 200.75.231.230:32778 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-d87543-9624df6f75622b0a-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=ee67d338 To: "Jorge Oscar Pinzón" Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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/1000-5f91", "callprocessing|1018|SIP/1018") in new stack -- Executing Set("SIP/1000-5f91", "LANGUAGE()=es") in new stack -- Executing Set("SIP/1000-5f91", "DB(LastCalledExt/1000)=1018") in new stack -- Executing Set("SIP/1000-5f91", "DB(LastCallingExt/1018)=1000") in new stack -- Executing Dial("SIP/1000-5f91", "SIP/1018|30") in new stack We're at 200.75.200.134 port 15438 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=z9hG4bK7054b3ad From: "Irving Barría" ;tag=as2215dd61 To: Contact: Call-ID: 742356d4416413f32146aa112ee6ee66@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 16:56:48 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 15438 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=z9hG4bK7054b3ad Contact: To: ;tag=7b3f5d5a From: "Irving Barría";tag=as2215dd61 Call-ID: 742356d4416413f32146aa112ee6ee66@200.75.200.134 CSeq: 102 INVITE User-Agent: X-Lite release 1002tx stamp 29712 Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/1018-becc 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-d87543-9624df6f75622b0a-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=ee67d338 To: "Jorge Oscar Pinzón";tag=as16b8fd75 Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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=z9hG4bK7054b3ad Contact: To: ;tag=7b3f5d5a From: "Irving Barría";tag=as2215dd61 Call-ID: 742356d4416413f32146aa112ee6ee66@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=- 0 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 51160 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:F7A1C71226D3409DAB16E90B2105DC5E --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:51160 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=z9hG4bK1d7bcc1b From: "Irving Barría" ;tag=as2215dd61 To: ;tag=7b3f5d5a Contact: Call-ID: 742356d4416413f32146aa112ee6ee66@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-becc answered SIP/1000-5f91 We're at 200.75.200.134 port 15332 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-d87543-9624df6f75622b0a-1--d87543-;rport;received=200.75.231.230 From: "Irving Barría";tag=ee67d338 To: "Jorge Oscar Pinzón";tag=as16b8fd75 Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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 15332 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/1000-5f91 and SIP/1018-becc 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 15438 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=z9hG4bK37e1bb96 From: "Irving Barría" ;tag=as2215dd61 To: ;tag=7b3f5d5a Contact: Call-ID: 742356d4416413f32146aa112ee6ee66@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 32778 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 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK7054b3ad Contact: To: ;tag=7b3f5d5a From: "Irving Barría";tag=as2215dd61 Call-ID: 742356d4416413f32146aa112ee6ee66@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=- 0 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 51160 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:F7A1C71226D3409DAB16E90B2105DC5E --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:51160 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=z9hG4bK01ff2a60 From: "Irving Barría" ;tag=as2215dd61 To: ;tag=7b3f5d5a Contact: Call-ID: 742356d4416413f32146aa112ee6ee66@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 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-c4267446cf082e1f-1--d87543-;rport Max-Forwards: 70 Contact: To: "Jorge Oscar Pinzón";tag=as16b8fd75 From: "Irving Barría";tag=ee67d338 Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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 15332 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=z9hG4bK219f858d From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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 51160 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 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK37e1bb96 Contact: To: ;tag=7b3f5d5a From: "Irving Barría";tag=as2215dd61 Call-ID: 742356d4416413f32146aa112ee6ee66@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=- 0 2 IN IP4 201.221.243.70 s= c=IN IP4 201.221.243.70 t=0 0 m=audio 51160 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:F7A1C71226D3409DAB16E90B2105DC5E --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 201.221.243.70:51160 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=z9hG4bK50984511 From: "Irving Barría" ;tag=as2215dd61 To: ;tag=7b3f5d5a Contact: Call-ID: 742356d4416413f32146aa112ee6ee66@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 200.75.231.230:31878: SIP/2.0 200 OK Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK219f858d Contact: To: "Irving Barría";tag=ee67d338 From: "Jorge Oscar Pinzón";tag=as16b8fd75 Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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=- 0 3 IN IP4 200.75.231.230 s= c=IN IP4 200.75.231.230 t=0 0 m=audio 32778 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:1551168D3C8143148392210A0FE6342D --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 200.75.231.230:32778 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=z9hG4bK6fd0869e From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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: BYE sip:1000@200.75.200.134 SIP/2.0 Via: SIP/2.0/UDP 201.221.243.70:34000;branch=z9hG4bK-d87543-c84a557f757b1411-1--d87543-;rport Max-Forwards: 70 Contact: To: "Irving Barría";tag=as2215dd61 From: ;tag=7b3f5d5a Call-ID: 742356d4416413f32146aa112ee6ee66@200.75.200.134 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 201.221.243.70 : 34000 (NAT) Transmitting (NAT) to 201.221.243.70:34000: SIP/2.0 200 OK Via: SIP/2.0/UDP 201.221.243.70:34000;branch=z9hG4bK-d87543-c84a557f757b1411-1--d87543-;rport;received=201.221.243.70 From: ;tag=7b3f5d5a To: "Irving Barría";tag=as2215dd61 Call-ID: 742356d4416413f32146aa112ee6ee66@200.75.200.134 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 200.75.231.230, port 40778 We're at 200.75.200.134 port 15332 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=z9hG4bK06037613 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 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 20149 IN IP4 200.75.200.134 s=session c=IN IP4 200.75.200.134 t=0 0 m=audio 15332 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/1000-5f91' in macro 'callprocessing' == Spawn extension (macro-callprocessing, s, 4) exited non-zero on 'SIP/1000-5f91' --- 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=z9hG4bK06037613 Contact: To: "Irving Barría";tag=ee67d338 From: "Jorge Oscar Pinzón";tag=as16b8fd75 Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. 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=- 0 3 IN IP4 200.75.231.230 s= c=IN IP4 200.75.231.230 t=0 0 m=audio 32778 RTP/AVP 3 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:1551168D3C8143148392210A0FE6342D --- (11 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 200.75.231.230:32778 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 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=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 ACK User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 200.75.231.230, port 40778 Reliably Transmitting (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ.' in 32000 ms Destroying call '742356d4416413f32146aa112ee6ee66@200.75.200.134' --- Retransmitting #1 (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Retransmitting #2 (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Retransmitting #3 (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Retransmitting #4 (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Retransmitting #5 (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Retransmitting #6 (NAT) to 200.75.231.230:31878: CANCEL sip:1000@200.75.231.230:40778 SIP/2.0 Via: SIP/2.0/UDP 200.75.200.134:5060;branch=z9hG4bK0f926655 From: "Jorge Oscar Pinzón";tag=as16b8fd75 To: "Irving Barría";tag=ee67d338 Contact: Call-ID: 5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. CSeq: 103 CANCEL User-Agent: Nimbus SIP Server v2.00.1 Max-Forwards: 70 Content-Length: 0 --- Jul 12 11:57:34 WARNING[20157]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 5454664a77447860NWU0ZDYxNGQxYz VlOTk3MTZmMjM0NjJhNGU2YWJhNGQ. for seqno 103 (Non-critical Request) Destroying call '5454664a77447860NWU0ZDYxNGQxYzVlOTk3MTZmMjM0NjJhNGU2YWJhNGQ.'