<--- SIP read from UDP://10.10.10.197:5060 ---> INVITE sip:089262415703@10.10.10.207;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-b7a6407283fed535-1---d8754z- Max-Forwards: 70 Contact: To: From: ;tag=6229e56f Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6751 Content-Length: 331 v=0 o=Zoiper_user 0 0 IN IP4 10.10.10.197 s=Zoiper_session c=IN IP4 10.10.10.197 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (12 headers 15 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Sending to 10.10.10.197 : 5060 (no NAT) Using INVITE request as basis request - NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. Found user '1010' for '1010' <--- Reliably Transmitting (no NAT) to 10.10.10.197:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-b7a6407283fed535-1---d8754z-;received=10.10.10.197 From: ;tag=6229e56f To: ;tag=as519c2249 Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 1 INVITE User-Agent: WC SoftSwitch 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip.westcall.ru", nonce="56e74d3a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY.' in 32000 ms (Method: INVITE) <--- SIP read from UDP://10.10.10.197:5060 ---> ACK sip:089262415703@10.10.10.207;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-b7a6407283fed535-1---d8754z- Max-Forwards: 70 To: ;tag=as519c2249 From: ;tag=6229e56f Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP://10.10.10.197:5060 ---> INVITE sip:089262415703@10.10.10.207;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-5da0975de715414f-1---d8754z- Max-Forwards: 70 Contact: To: From: ;tag=6229e56f Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp User-Agent: Zoiper rev.6751 Authorization: Digest username="1010",realm="sip.westcall.ru",nonce="56e74d3a",uri="sip:089262415703@10.10.10.207;transport=UDP",response="eb8e476e6f9188c4232d44dc5467499b",algorithm=MD5 Content-Length: 331 v=0 o=Zoiper_user 0 0 IN IP4 10.10.10.197 s=Zoiper_session c=IN IP4 10.10.10.197 t=0 0 m=audio 8000 RTP/AVP 3 0 8 110 98 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 15 lines) --- Sending to 10.10.10.197 : 5060 (no NAT) Using INVITE request as basis request - NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. Found user '1010' for '1010' Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 110 Found RTP audio format 98 Found RTP audio format 101 Found audio description format GSM for ID 3 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format speex for ID 110 Found audio description format iLBC for ID 98 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.10.197:8000 Looking for 089262415703 in default-fax (domain 10.10.10.207) list_route: hop: <--- Transmitting (no NAT) to 10.10.10.197:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-5da0975de715414f-1---d8754z-;received=10.10.10.197 From: ;tag=6229e56f To: Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 2 INVITE User-Agent: WC SoftSwitch 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing DIAL("SIP/1010-00000006", "SIP/voipgw4/09689262415703") == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 == Using UDPTL CoS mark 5 Audio is at 10.10.10.207 port 12978 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.10.219:5060: INVITE sip:09689262415703@10.10.10.219:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK29fee4d6;rport Max-Forwards: 70 From: "4995019920" ;tag=as4d186045 To: Contact: Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 CSeq: 102 INVITE User-Agent: WC SoftSwitch 2.1 Remote-Party-ID: "4995019920" ;privacy=off;screen=no Date: Fri, 16 Apr 2010 10:07:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 289 v=0 o=wc_ss_v21 86610080 86610080 IN IP4 10.10.10.207 s=WC SoftSwitch 2.1 c=IN IP4 10.10.10.207 t=0 0 m=audio 12978 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called voipgw4/09689262415703 <--- SIP read from UDP://10.10.10.219:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK29fee4d6;rport From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Date: Fri, 16 Apr 2010 10:07:50 GMT Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP://10.10.10.197:59810 ---> <-------------> <--- SIP read from UDP://10.10.10.219:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK29fee4d6;rport From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Date: Fri, 16 Apr 2010 10:07:50 GMT Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 252 v=0 o=CiscoSystemsSIP-GW-UserAgent 613 9995 IN IP4 10.10.10.219 s=SIP Call c=IN IP4 10.10.10.219 t=0 0 m=audio 16688 RTP/AVP 8 101 c=IN IP4 10.10.10.219 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.10.10.219:16688 -- SIP/voipgw4-00000007 is making progress passing it to SIP/1010-00000006 Audio is at 10.10.10.207 port 15650 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to 10.10.10.197:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-5da0975de715414f-1---d8754z-;received=10.10.10.197 From: ;tag=6229e56f To: ;tag=as76603291 Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 2 INVITE User-Agent: WC SoftSwitch 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=wc_ss_v21 594103337 594103337 IN IP4 10.10.10.207 s=WC SoftSwitch 2.1 c=IN IP4 10.10.10.207 t=0 0 m=audio 15650 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP://10.10.10.219:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK29fee4d6;rport From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Date: Fri, 16 Apr 2010 10:07:50 GMT Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 252 v=0 o=CiscoSystemsSIP-GW-UserAgent 613 9995 IN IP4 10.10.10.219 s=SIP Call c=IN IP4 10.10.10.219 t=0 0 m=audio 16688 RTP/AVP 8 101 c=IN IP4 10.10.10.219 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- -- SIP/voipgw4-00000007 is making progress passing it to SIP/1010-00000006 <--- SIP read from UDP://10.10.10.219:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK29fee4d6;rport From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Date: Fri, 16 Apr 2010 10:07:50 GMT Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Supported: replaces Allow-Events: telephone-event Contact: Content-Type: application/sdp Content-Length: 252 v=0 o=CiscoSystemsSIP-GW-UserAgent 613 9995 IN IP4 10.10.10.219 s=SIP Call c=IN IP4 10.10.10.219 t=0 0 m=audio 16688 RTP/AVP 8 101 c=IN IP4 10.10.10.219 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 <-------------> --- (14 headers 11 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.219, port 5060 Transmitting (NAT) to 10.10.10.219:5060: ACK sip:09689262415703@10.10.10.219:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK4d50d2a2;rport Max-Forwards: 70 From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Contact: Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 CSeq: 102 ACK User-Agent: WC SoftSwitch 2.1 Remote-Party-ID: "4995019920" ;privacy=off;screen=no Content-Length: 0 --- -- SIP/voipgw4-00000007 answered SIP/1010-00000006 Audio is at 10.10.10.207 port 15650 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.10.10.197:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-5da0975de715414f-1---d8754z-;received=10.10.10.197 From: ;tag=6229e56f To: ;tag=as76603291 Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 2 INVITE User-Agent: WC SoftSwitch 2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 291 v=0 o=wc_ss_v21 594103337 594103338 IN IP4 10.10.10.207 s=WC SoftSwitch 2.1 c=IN IP4 10.10.10.207 t=0 0 m=audio 15650 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/1010-00000006 and SIP/voipgw4-00000007 <--- SIP read from UDP://10.10.10.197:5060 ---> ACK sip:089262415703@10.10.10.207 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.197:5060;branch=z9hG4bK-d8754z-f671dbf210621eeb-1---d8754z- Max-Forwards: 70 Contact: To: ;tag=as76603291 From: ;tag=6229e56f Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 2 ACK User-Agent: Zoiper rev.6751 Authorization: Digest username="1010",realm="sip.westcall.ru",nonce="56e74d3a",uri="sip:089262415703@10.10.10.207;transport=UDP",response="eb8e476e6f9188c4232d44dc5467499b",algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP://10.10.10.197:5060 ---> <-------------> [Apr 16 14:08:18] NOTICE[21771]: chan_sip.c:20129 check_rtp_timeout: Disconnecting call 'SIP/voipgw4-00000007' for lack of RTP activity in 18 seconds Scheduling destruction of SIP dialog '65d1967d5145214936a4687e5f318d19@10.10.10.207' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.219, port 5060 Reliably Transmitting (NAT) to 10.10.10.219:5060: BYE sip:09689262415703@10.10.10.219:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK74fab037;rport Max-Forwards: 70 From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 CSeq: 103 BYE User-Agent: WC SoftSwitch 2.1 Remote-Party-ID: "4995019920" ;privacy=off;screen=no X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (default-fax, 089262415703, 2) exited non-zero on 'SIP/1010-00000006' Scheduling destruction of SIP dialog 'NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY.' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.197, port 5060 Reliably Transmitting (no NAT) to 10.10.10.197:5060: BYE sip:1010@10.10.10.197:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK42d6059a;rport Max-Forwards: 70 From: ;tag=as76603291 To: ;tag=6229e56f Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 102 BYE User-Agent: WC SoftSwitch 2.1 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP://10.10.10.219:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK74fab037;rport From: "4995019920" ;tag=as4d186045 To: ;tag=76988420-26B9 Date: Fri, 16 Apr 2010 10:08:19 GMT Call-ID: 65d1967d5145214936a4687e5f318d19@10.10.10.207 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 103 BYE <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '65d1967d5145214936a4687e5f318d19@10.10.10.207' Method: INVITE <--- SIP read from UDP://10.10.10.197:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.207:5060;branch=z9hG4bK42d6059a;rport=5060 Contact: To: ;tag=6229e56f From: ;tag=as76603291 Call-ID: NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY. CSeq: 102 BYE User-Agent: Zoiper rev.6751 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'NDViMTVjOTk2YWY5NjM4MWZjNmFiMDFlZGQ1YWQ2MWY.' Method: ACK