localhost*CLI> <--- SIP read from 189.180.217.155:55977 ---> INVITE sip:0016044690084@sip.telavip.com.mx SIP/2.0 Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-add2164cc95d1441-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;transport=UDP;tag=d813117c Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp User-Agent: Zoiper rev.417 Content-Length: 250 v=0 o= 0 0 IN IP4 189.180.217.155 s= c=IN IP4 189.180.217.155 t=0 0 m=audio 56038 RTP/AVP 3 110 97 101 a=fmtp:97 mode=30 a=fmtp:101 0-15 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (12 headers 12 lines) --- Sending to 189.180.217.155 : 55977 (NAT) Using INVITE request as basis request - ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. <--- Reliably Transmitting (no NAT) to 189.180.217.155:55828 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-add2164cc95d1441-1---d8754z-;rport;received=189.180.217.155 From: ;transport=UDP;tag=d813117c To: ;tag=as3066808d Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c389d18" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.' in 32000 ms (Method: INVITE) Found user '5512047763' localhost*CLI> <--- SIP read from 189.180.217.155:55977 ---> ACK sip:0016044690084@sip.telavip.com.mx SIP/2.0 Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-add2164cc95d1441-1---d8754z-;rport To: ;tag=as3066808d From: ;transport=UDP;tag=d813117c Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- localhost*CLI> <--- SIP read from 189.180.217.155:55977 ---> INVITE sip:0016044690084@sip.telavip.com.mx SIP/2.0 Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;transport=UDP;tag=d813117c Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Type: application/sdp Proxy-Authorization: Digest username="5512047763",realm="asterisk",nonce="7c389d18",uri="sip:0016044690084@sip.telavip.com.mx",response="c4315cb52245bc9c43111d2b11281f12",algorithm=MD5 User-Agent: Zoiper rev.417 Content-Length: 250 v=0 o= 0 0 IN IP4 189.180.217.155 s= c=IN IP4 189.180.217.155 t=0 0 m=audio 56038 RTP/AVP 3 110 97 101 a=fmtp:97 mode=30 a=fmtp:101 0-15 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (13 headers 12 lines) --- Sending to 189.180.217.155 : 55977 (NAT) Using INVITE request as basis request - ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. Found user '5512047763' Found RTP audio format 3 Found RTP audio format 110 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 189.180.217.155:56038 Found description format GSM for ID 3 Found description format speex for ID 110 Found description format iLBC for ID 97 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x602 (gsm|speex|ilbc)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 189.180.217.155:56038 Looking for 0016044690084 in salida-df (domain sip.telavip.com.mx) list_route: hop: <--- Transmitting (no NAT) to 189.180.217.155:55828 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport;received=189.180.217.155 From: ;transport=UDP;tag=d813117c To: Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [0016044690084@salida-df:1] Goto("SIP/5512047763-08b1b2b0", "salida-internacional|0016044690084|1") in new stack -- Goto (salida-internacional,0016044690084,1) -- Executing [0016044690084@salida-internacional:1] Dial("SIP/5512047763-08b1b2b0", "SIP/16044690084@sip.ipacomm.net|30|r") in new stack Audio is at 200.76.98.137 port 11802 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 208.239.76.163:5060: INVITE sip:16044690084@sip.ipacomm.net SIP/2.0 Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK58a79278;rport From: "Usuario Prueba" ;tag=as0c351c86 To: Contact: Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 20 Nov 2007 00:51:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 19922 19922 IN IP4 200.76.98.137 s=session c=IN IP4 200.76.98.137 t=0 0 m=audio 11802 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 16044690084@sip.ipacomm.net <--- Transmitting (no NAT) to 189.180.217.155:55828 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport;received=189.180.217.155 From: ;transport=UDP;tag=d813117c To: ;tag=as0aba4a06 Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> localhost*CLI> <--- SIP read from 208.239.76.163:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.76.98.137:5060;received=200.76.98.137;branch=z9hG4bK58a79278;rport=5060 From: "Usuario Prueba" ;tag=as0c351c86 To: Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137 CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- localhost*CLI> <--- SIP read from 208.239.76.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 200.76.98.137:5060;received=200.76.98.137;branch=z9hG4bK58a79278;rport=5060 From: "Usuario Prueba" ;tag=as0c351c86 To: ;tag=SD5i84f99-e3736a12 Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137 CSeq: 102 INVITE X-UUID: a60162e825054ffe8d5c0e63f7f2c53f Contact: Content-Length: 267 ontent-Type: application/sdp Allow: INVITE v=0 o=root 29482 29482 IN IP4 208.239.76.163 s=session c=IN IP4 208.239.76.163 t=0 0 m=audio 47970 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - <-------------> --- (11 headers 12 lines) --- Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 208.239.76.163:47970 Found description format GSM for ID 3 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 208.239.76.163:47970 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 208.239.76.163, port 5060 Transmitting (no NAT) to 208.239.76.163:5060: ACK sip:16044690084@208.239.76.163:5060;wlsscid=1ae4691c270666;transport=udp SIP/2.0 Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK54f9419e;rport From: "Usuario Prueba" ;tag=as0c351c86 To: ;tag=SD5i84f99-e3736a12 Contact: Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/sip.ipacomm.net-08bb2a30 answered SIP/5512047763-08b1b2b0 Audio is at 200.76.98.137 port 16566 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 189.180.217.155:55828 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-4fe4cc06ff31530d-1---d8754z-;rport;received=189.180.217.155 From: ;transport=UDP;tag=d813117c To: ;tag=as0aba4a06 Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 19922 19922 IN IP4 200.76.98.137 s=session c=IN IP4 200.76.98.137 t=0 0 m=audio 16566 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/5512047763-08b1b2b0 and SIP/sip.ipacomm.net-08bb2a30 localhost*CLI> <--- SIP read from 189.180.217.155:55977 ---> ACK sip:0016044690084@200.76.98.137 SIP/2.0 Via: SIP/2.0/UDP 189.180.217.155:55828;branch=z9hG4bK-d8754z-60c1e71a0d6209b7-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as0aba4a06 From: ;transport=UDP;tag=d813117c Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 2 ACK Proxy-Authorization: Digest username="5512047763",realm="asterisk",nonce="7c389d18",uri="sip:0016044690084@sip.telavip.com.mx",response="c4315cb52245bc9c43111d2b11281f12",algorithm=MD5 User-Agent: Zoiper rev.417 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- localhost*CLI> <--- SIP read from 208.239.76.163:5060 ---> BYE sip:5512047763@200.76.98.137 SIP/2.0 Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bKu9fupa104o0glf4th7g1sd0000g00.1 To: "Usuario Prueba" ;tag=as0c351c86 Content-Length: 0 CSeq: 1 BYE Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137 From: ;tag=SD5i84f99-e3736a12 Max-Forwards: 68 <-------------> --- (8 headers 0 lines) --- Sending to 208.239.76.163 : 5060 (no NAT) <--- Transmitting (no NAT) to 208.239.76.163:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 208.239.76.163:5060;branch=z9hG4bKu9fupa104o0glf4th7g1sd0000g00.1;received=208.239.76.163 From: ;tag=SD5i84f99-e3736a12 To: "Usuario Prueba" ;tag=as0c351c86 Call-ID: 4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (salida-internacional, 0016044690084, 1) exited non-zero on 'SIP/5512047763-08b1b2b0' Scheduling destruction of SIP dialog 'ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 189.180.217.155, port 55828 Reliably Transmitting (no NAT) to 189.180.217.155:55828: BYE sip:5512047763@189.180.217.155:55828 SIP/2.0 Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK3a9c95bf From: ;tag=as0aba4a06 To: ;transport=UDP;tag=d813117c Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '4e14d2eb06413e2523ba9dbe100e4981@200.76.98.137' Method: BYE localhost*CLI> <--- SIP read from 189.180.217.155:55977 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 200.76.98.137:5060;branch=z9hG4bK3a9c95bf Contact: To: ;transport=UDP;tag=d813117c From: ;tag=as0aba4a06 Call-ID: ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk. CSeq: 102 BYE User-Agent: Zoiper rev.417 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'ZWJiYmRiNDEwMzIxMTE3NGQ0MWM2YWMxMTNlMjNiZTk.' Method: ACK localhost*CLI> sip <--- SIP read from 189.180.217.155:55977 ---> <-------------> --- (0 headers 1 lines) ---