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ShadowAnt

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  1. ShadowAnt
    I try it on 32-bit edition of windows server 2003, windows XP, but still have the same problem...
  2. ShadowAnt
    The host have a static IP. maybe it is because windows 2008 x64...?
  3. ShadowAnt
    <port_bind4/><port_bind6/> I didn't see any parameters in this string...
  4. ShadowAnt
    Hello! I install faxback and I have a trouble with recieving faxes. I try to send fax from analog fax-sip provider (with T.38) - pbxnsip - NET SatisFAXtion here is a log: 1.1.1.3 - faxback 1.1.1.5 - pbxnsip 1.1.1.101 - sip provider [7] 2010/02/05 08:28:11: Last message repeated 2 times [6] 2010/02/05 08:28:11: Received DTMF F [7] 2010/02/05 08:28:11: Attendant: Calling extension 777 [7] 2010/02/05 08:28:11: SIP Tx udp:1.1.1.3:5060: INVITE sip:777@1.1.1.3:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru> Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Max-Forwards: 70 Contact: <sip:777@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 327 v=0 o=- 13668 13668 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=audio 63512 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2010/02/05 08:28:11: SIP Rx udp:1.1.1.3:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Contact: <sip:IPFax@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [7] 2010/02/05 08:28:11: SIP Rx udp:1.1.1.3:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Contact: <sip:IPFax@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [7] 2010/02/05 08:28:11: SIP Rx udp:1.1.1.3:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Contact: <sip:IPFax@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 210 v=0 o=IPFax 0 0 IN IP4 1.1.1.3 s=SIP Fax Call i=IPFax c=IN IP4 1.1.1.3 t=0 0 m=audio 49248 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [7] 2010/02/05 08:28:11: Call 6ec1ef94@pbx#5943: Clear last INVITE [7] 2010/02/05 08:28:11: Set packet length to 20 [6] 2010/02/05 08:28:11: Send codec=pcmu/8000 afrer answer [6] 2010/02/05 08:28:11: Sending RTP for 6ec1ef94@pbx#5943 to 1.1.1.3:49248 [7] 2010/02/05 08:28:11: SIP Tx udp:1.1.1.3:5060: ACK sip:IPFax@1.1.1.3:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-37ac123baf4fee949d4eaf868c72d713;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 ACK Max-Forwards: 70 Contact: <sip:777@1.1.1.5:5060;transport=udp> Content-Length: 0 [7] 2010/02/05 08:28:11: Determine pass-through mode after receiving response [7] 2010/02/05 08:28:11: 6ec1ef94@pbx#5943: RTP pass-through mode [7] 2010/02/05 08:28:11: 7007803330339123745-1265347685@172.30.77.25#f72475646e: RTP pass-through mode [7] 2010/02/05 08:28:11: 7007803330339123745-1265347685@172.30.77.25#f72475646e: Media-aware pass-through mode [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.3:5060: INVITE sip:8632370680@pbx.case.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 INVITE Max-Forwards: 70 Contact: <sip:777@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Session-Expires: 3600;refresher=uas Supported: timer,replaces,billing,presence,* Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 347 v=0 o=IPFax 0 1 IN IP4 1.1.1.3 s=SIP Fax Call i=IPFax c=IN IP4 1.1.1.3 t=0 0 m=image 49200 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxUdpEC:t38UDPRedundancy [0] 2010/02/05 08:28:15: UDP: bind() to port 54900 failed [7] 2010/02/05 08:28:15: UDP: Opening socket on :57948 [7] 2010/02/05 08:28:15: UDP: Opening socket on :57140 [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.101:5060: INVITE sip:8632370680@1.1.1.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-290b90ac185fe1cdcaf1380f07c14040;rport From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 INVITE Max-Forwards: 70 Contact: <sip:2688634@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 331 v=0 o=- 54151 54152 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=image 57140 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxUdpEC:t38UDPRedundancy [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.3:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 INVITE Content-Length: 0 [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.5:5060;received=1.1.1.5;branch=z9hG4bK-290b90ac185fe1cdcaf1380f07c14040;rport=5060 From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 INVITE [5] 2010/02/05 08:28:15: Passthrough: Changing destination [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;received=1.1.1.5;branch=z9hG4bK-290b90ac185fe1cdcaf1380f07c14040;rport=5060 From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 INVITE Content-Type: application/sdp Contact: <sip:8632370680@1.1.1.101:5060;transport=udp>;user=phone Supported: timer,100rel Content-Length: 103 v=0 o=MG4000|2.0 3420 6706 IN IP4 1.1.1.101 s=- c=IN IP4 1.1.1.101 t=0 0 m=image 52860 udptl t38 [7] 2010/02/05 08:28:15: Call 7007803330339123745-1265347685@172.30.77.25#f72475646e: Clear last INVITE [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.3:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 INVITE Contact: <sip:777@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 92 v=0 o=- 13668 13669 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=image 57948 udptl t38 [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.101:5060: ACK sip:8632370680@1.1.1.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-91155c8a0f5db5c3c077e16dd28ac96c;rport From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 ACK Max-Forwards: 70 Contact: <sip:2688634@1.1.1.5:5060;transport=udp> Content-Length: 0 [7] 2010/02/05 08:28:15: Determine pass-through mode after receiving response [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.3:5060: ACK sip:777@1.1.1.5:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 ACK Max-Forwards: 70 User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [5] 2010/02/05 08:28:26: SIP port accept from 1.1.7.8:63891 [7] 2010/02/05 08:28:26: SIP Rx tcp:1.1.7.8:63891: OPTIONS sip:1.1.1.5:5060 SIP/2.0 FROM: <sip:s-caseexch.case.ru:5060;transport=Tcp;ms-opaque=2cb7e09927aa4652>;epid=FC0ADCD96D;tag=b9a0d84671 TO: <sip:1.1.1.5:5060> CSEQ: 18599 OPTIONS CALL-ID: b819daf2a97c490ebf7d55910cc06236 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 1.1.7.8:63891;branch=z9hG4bK27c8619a ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.1.0.0 [7] 2010/02/05 08:28:26: SIP Tx tcp:1.1.7.8:63891: SIP/2.0 200 Ok Via: SIP/2.0/TCP 1.1.7.8:63891;branch=z9hG4bK27c8619a From: <sip:s-caseexch.case.ru:5060;transport=Tcp;ms-opaque=2cb7e09927aa4652>;tag=b9a0d84671;epid=FC0ADCD96D To: <sip:1.1.1.5:5060>;tag=f7ce52c8fc Call-ID: b819daf2a97c490ebf7d55910cc06236 CSeq: 18599 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [7] 2010/02/05 08:28:30: SIP Rx udp:1.1.1.101:5060: INVITE sip:2688634@1.1.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 Call-ID: 7007803330339123745-1265347685@172.30.77.25 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Content-Type: application/sdp CSeq: 2 INVITE Contact: <sip:8632370680@1.1.1.101:5060;transport=udp>;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO Supported: timer,100rel Max-Forwards: 69 Content-Length: 222 v=0 o=MG4000|2.0 3420 6707 IN IP4 1.1.1.101 s=- c=IN IP4 1.1.1.101 t=0 0 m=audio 52860 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 a=rtpmap:13 CN/8000 [7] 2010/02/05 08:28:30: Set packet length to 20 [7] 2010/02/05 08:28:30: SIP Tx udp:1.1.1.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 2 INVITE Content-Length: 0 [6] 2010/02/05 08:28:41: SIP TCP/TLS timeout on 1.1.1.4:41773, closing connection [7] 2010/02/05 08:28:57: SIP Rx udp:1.1.1.3:5060: BYE sip:777@1.1.1.5:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1012 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21704 BYE Max-Forwards: 70 User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [7] 2010/02/05 08:28:57: SIP Tx udp:1.1.1.3:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1012 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21704 BYE Contact: <sip:777@1.1.1.5:5060;transport=udp> User-Agent: pbxnsip-PBX/3.4.0.3201 RTP-RxStat: Dur=46,Pkt=734,Oct=43912,Underun=0 RTP-TxStat: Dur=46,Pkt=910,Oct=155327 Content-Length: 0 [7] 2010/02/05 08:28:57: Other Ports: 1 [7] 2010/02/05 08:28:57: Call Port: 7007803330339123745-1265347685@172.30.77.25#f72475646e [7] 2010/02/05 08:28:57: SIP Tx udp:1.1.1.101:5060: BYE sip:8632370680@1.1.1.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-8c7f32049f3c81d47cc43e0eea58e3a2;rport From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27404 BYE Max-Forwards: 70 Contact: <sip:2688634@1.1.1.5:5060;transport=udp> RTP-RxStat: Dur=52,Pkt=1210,Oct=206927,Underun=0 RTP-TxStat: Dur=42,Pkt=3162,Oct=461528 Content-Length: 0 [7] 2010/02/05 08:28:57: SIP Rx udp:1.1.1.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;received=1.1.1.5;branch=z9hG4bK-8c7f32049f3c81d47cc43e0eea58e3a2;rport=5060 From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27404 BYE Contact: <sip:8632370680@1.1.1.101:5060;transport=udp>;user=phone Supported: timer,100rel Content-Length: 0 [7] 2010/02/05 08:28:57: Call 7007803330339123745-1265347685@172.30.77.25#f72475646e: Clear last request [5] 2010/02/05 08:28:57: BYE Response: Terminate 7007803330339123745-1265347685@172.30.77.25 [7] 2010/02/05 08:29:02: SIP Rx udp:1.1.1.101:5060: CANCEL sip:2688634@1.1.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 CSeq: 2 CANCEL Call-ID: 7007803330339123745-1265347685@172.30.77.25 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Max-Forwards: 69 Content-Length: 0 [7] 2010/02/05 08:29:02: SIP Tx udp:1.1.1.101:5060: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 2 CANCEL Content-Length: 0
  5. ShadowAnt
    Hello! I install faxback and I have a trouble with recieving faxes. I try to send fax from analog fax-sip provider (with T.38) - pbxnsip - NET SatisFAXtion here is a log: 1.1.1.3 - faxback 1.1.1.5 - pbxnsip 1.1.1.101 - sip provider [7] 2010/02/05 08:28:11: Last message repeated 2 times [6] 2010/02/05 08:28:11: Received DTMF F [7] 2010/02/05 08:28:11: Attendant: Calling extension 777 [7] 2010/02/05 08:28:11: SIP Tx udp:1.1.1.3:5060: INVITE sip:777@1.1.1.3:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru> Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Max-Forwards: 70 Contact: <sip:777@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Alert-Info: <http://127.0.0.1/Bellcore-dr3> Content-Type: application/sdp Content-Length: 327 v=0 o=- 13668 13668 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=audio 63512 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2010/02/05 08:28:11: SIP Rx udp:1.1.1.3:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Contact: <sip:IPFax@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [7] 2010/02/05 08:28:11: SIP Rx udp:1.1.1.3:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Contact: <sip:IPFax@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [7] 2010/02/05 08:28:11: SIP Rx udp:1.1.1.3:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-77d89e821eab1998222b24bcb014ca34;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 INVITE Contact: <sip:IPFax@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 210 v=0 o=IPFax 0 0 IN IP4 1.1.1.3 s=SIP Fax Call i=IPFax c=IN IP4 1.1.1.3 t=0 0 m=audio 49248 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv [7] 2010/02/05 08:28:11: Call 6ec1ef94@pbx#5943: Clear last INVITE [7] 2010/02/05 08:28:11: Set packet length to 20 [6] 2010/02/05 08:28:11: Send codec=pcmu/8000 afrer answer [6] 2010/02/05 08:28:11: Sending RTP for 6ec1ef94@pbx#5943 to 1.1.1.3:49248 [7] 2010/02/05 08:28:11: SIP Tx udp:1.1.1.3:5060: ACK sip:IPFax@1.1.1.3:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-37ac123baf4fee949d4eaf868c72d713;rport From: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 To: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 Call-ID: 6ec1ef94@pbx CSeq: 21702 ACK Max-Forwards: 70 Contact: <sip:777@1.1.1.5:5060;transport=udp> Content-Length: 0 [7] 2010/02/05 08:28:11: Determine pass-through mode after receiving response [7] 2010/02/05 08:28:11: 6ec1ef94@pbx#5943: RTP pass-through mode [7] 2010/02/05 08:28:11: 7007803330339123745-1265347685@172.30.77.25#f72475646e: RTP pass-through mode [7] 2010/02/05 08:28:11: 7007803330339123745-1265347685@172.30.77.25#f72475646e: Media-aware pass-through mode [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.3:5060: INVITE sip:8632370680@pbx.case.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 INVITE Max-Forwards: 70 Contact: <sip:777@1.1.1.3:5060> User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Session-Expires: 3600;refresher=uas Supported: timer,replaces,billing,presence,* Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY Content-Type: application/sdp Content-Length: 347 v=0 o=IPFax 0 1 IN IP4 1.1.1.3 s=SIP Fax Call i=IPFax c=IN IP4 1.1.1.3 t=0 0 m=image 49200 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxUdpEC:t38UDPRedundancy [0] 2010/02/05 08:28:15: UDP: bind() to port 54900 failed [7] 2010/02/05 08:28:15: UDP: Opening socket on :57948 [7] 2010/02/05 08:28:15: UDP: Opening socket on :57140 [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.101:5060: INVITE sip:8632370680@1.1.1.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-290b90ac185fe1cdcaf1380f07c14040;rport From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 INVITE Max-Forwards: 70 Contact: <sip:2688634@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 331 v=0 o=- 54151 54152 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=image 57140 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxUdpEC:t38UDPRedundancy [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.3:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 INVITE Content-Length: 0 [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.5:5060;received=1.1.1.5;branch=z9hG4bK-290b90ac185fe1cdcaf1380f07c14040;rport=5060 From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 INVITE [5] 2010/02/05 08:28:15: Passthrough: Changing destination [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;received=1.1.1.5;branch=z9hG4bK-290b90ac185fe1cdcaf1380f07c14040;rport=5060 From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 INVITE Content-Type: application/sdp Contact: <sip:8632370680@1.1.1.101:5060;transport=udp>;user=phone Supported: timer,100rel Content-Length: 103 v=0 o=MG4000|2.0 3420 6706 IN IP4 1.1.1.101 s=- c=IN IP4 1.1.1.101 t=0 0 m=image 52860 udptl t38 [7] 2010/02/05 08:28:15: Call 7007803330339123745-1265347685@172.30.77.25#f72475646e: Clear last INVITE [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.3:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 INVITE Contact: <sip:777@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 92 v=0 o=- 13668 13669 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=image 57948 udptl t38 [7] 2010/02/05 08:28:15: SIP Tx udp:1.1.1.101:5060: ACK sip:8632370680@1.1.1.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-91155c8a0f5db5c3c077e16dd28ac96c;rport From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27403 ACK Max-Forwards: 70 Contact: <sip:2688634@1.1.1.5:5060;transport=udp> Content-Length: 0 [7] 2010/02/05 08:28:15: Determine pass-through mode after receiving response [7] 2010/02/05 08:28:15: SIP Rx udp:1.1.1.3:5060: ACK sip:777@1.1.1.5:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1011 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21703 ACK Max-Forwards: 70 User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [5] 2010/02/05 08:28:26: SIP port accept from 1.1.7.8:63891 [7] 2010/02/05 08:28:26: SIP Rx tcp:1.1.7.8:63891: OPTIONS sip:1.1.1.5:5060 SIP/2.0 FROM: <sip:s-caseexch.case.ru:5060;transport=Tcp;ms-opaque=2cb7e09927aa4652>;epid=FC0ADCD96D;tag=b9a0d84671 TO: <sip:1.1.1.5:5060> CSEQ: 18599 OPTIONS CALL-ID: b819daf2a97c490ebf7d55910cc06236 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 1.1.7.8:63891;branch=z9hG4bK27c8619a ACCEPT: application/sdp CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.1.0.0 [7] 2010/02/05 08:28:26: SIP Tx tcp:1.1.7.8:63891: SIP/2.0 200 Ok Via: SIP/2.0/TCP 1.1.7.8:63891;branch=z9hG4bK27c8619a From: <sip:s-caseexch.case.ru:5060;transport=Tcp;ms-opaque=2cb7e09927aa4652>;tag=b9a0d84671;epid=FC0ADCD96D To: <sip:1.1.1.5:5060>;tag=f7ce52c8fc Call-ID: b819daf2a97c490ebf7d55910cc06236 CSeq: 18599 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [7] 2010/02/05 08:28:30: SIP Rx udp:1.1.1.101:5060: INVITE sip:2688634@1.1.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 Call-ID: 7007803330339123745-1265347685@172.30.77.25 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Content-Type: application/sdp CSeq: 2 INVITE Contact: <sip:8632370680@1.1.1.101:5060;transport=udp>;user=phone Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO Supported: timer,100rel Max-Forwards: 69 Content-Length: 222 v=0 o=MG4000|2.0 3420 6707 IN IP4 1.1.1.101 s=- c=IN IP4 1.1.1.101 t=0 0 m=audio 52860 RTP/AVP 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 a=rtpmap:13 CN/8000 [7] 2010/02/05 08:28:30: Set packet length to 20 [7] 2010/02/05 08:28:30: SIP Tx udp:1.1.1.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 2 INVITE Content-Length: 0 [6] 2010/02/05 08:28:41: SIP TCP/TLS timeout on 1.1.1.4:41773, closing connection [7] 2010/02/05 08:28:57: SIP Rx udp:1.1.1.3:5060: BYE sip:777@1.1.1.5:5060 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1012 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21704 BYE Max-Forwards: 70 User-Agent: Net Satisfaxtion/IP_FAX-8.5.4225.929 Content-Length: 0 [7] 2010/02/05 08:28:57: SIP Tx udp:1.1.1.3:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 1.1.1.3:5060;branch=z9hG4bK1012 From: <sip:777@pbx.case.ru>;tag=IPF_PORT_0024_1010 To: <sip:8632370680@pbx.case.ru:5060;user=phone>;tag=5943 Call-ID: 6ec1ef94@pbx CSeq: 21704 BYE Contact: <sip:777@1.1.1.5:5060;transport=udp> User-Agent: pbxnsip-PBX/3.4.0.3201 RTP-RxStat: Dur=46,Pkt=734,Oct=43912,Underun=0 RTP-TxStat: Dur=46,Pkt=910,Oct=155327 Content-Length: 0 [7] 2010/02/05 08:28:57: Other Ports: 1 [7] 2010/02/05 08:28:57: Call Port: 7007803330339123745-1265347685@172.30.77.25#f72475646e [7] 2010/02/05 08:28:57: SIP Tx udp:1.1.1.101:5060: BYE sip:8632370680@1.1.1.101:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-8c7f32049f3c81d47cc43e0eea58e3a2;rport From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27404 BYE Max-Forwards: 70 Contact: <sip:2688634@1.1.1.5:5060;transport=udp> RTP-RxStat: Dur=52,Pkt=1210,Oct=206927,Underun=0 RTP-TxStat: Dur=42,Pkt=3162,Oct=461528 Content-Length: 0 [7] 2010/02/05 08:28:57: SIP Rx udp:1.1.1.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;received=1.1.1.5;branch=z9hG4bK-8c7f32049f3c81d47cc43e0eea58e3a2;rport=5060 From: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e To: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 27404 BYE Contact: <sip:8632370680@1.1.1.101:5060;transport=udp>;user=phone Supported: timer,100rel Content-Length: 0 [7] 2010/02/05 08:28:57: Call 7007803330339123745-1265347685@172.30.77.25#f72475646e: Clear last request [5] 2010/02/05 08:28:57: BYE Response: Terminate 7007803330339123745-1265347685@172.30.77.25 [7] 2010/02/05 08:29:02: SIP Rx udp:1.1.1.101:5060: CANCEL sip:2688634@1.1.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 CSeq: 2 CANCEL Call-ID: 7007803330339123745-1265347685@172.30.77.25 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Max-Forwards: 69 Content-Length: 0 [7] 2010/02/05 08:29:02: SIP Tx udp:1.1.1.101:5060: SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 1.1.1.101:5060;branch=z9hG4bK5jh8d610700gvec8h7g0.1 From: <sip:8632370680@1.1.1.101:5060;user=phone>;tag=26237 To: <sip:2688634@1.1.1.5:5060;user=phone>;tag=f72475646e Call-ID: 7007803330339123745-1265347685@172.30.77.25 CSeq: 2 CANCEL Content-Length: 0
  6. ShadowAnt
    I recreate trunk with sipnet.ru and call to the extension was successfull! I don't know, have sipnet made any changes in their configuration or not... So DTMF transfer is working now with sipnet.
  7. ShadowAnt
    My phone is Office Communicator 2007 R2 but I think, if I start to use something like SNOM it will work fine. With Skype to SIP I don't have these problems, only with sipnet.ru. Unfortunually I can't test hardware phone, because I don't have it...
  8. ShadowAnt
    Unfotunally, I can't call skype users from "skype to sip" Skype support says that it is not supported... But from my Skype I can dial my "skype to sip" number.
  9. ShadowAnt
    this is sipnet.ru. Our Russian provider... I want to dial extension number after I dial telephone number. They want SIP-INFO DTMF, but I find on PBXnSIP forum, that this is not supported...
  10. ShadowAnt
    Hello! How I can change DTMF method in PBXnSIP? when I try to dial extension - service provider do not transfer it. here is a log: [7] 2010/01/12 14:32:00: SIP Rx tcp:1.1.7.7:58486: INVITE sip:+78632370684@1.1.1.5;user=phone SIP/2.0 FROM: <sip:+78632688634100@s-case1c.case.ru;user=phone>;epid=77B47EEE06;tag=a6cc0a75 TO: <sip:+78632370684@1.1.1.5;user=phone> CSEQ: 113 INVITE CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1 CONTACT: <sip:s-case1c.case.ru:5060;transport=Tcp;maddr=1.1.7.7;ms-opaque=d799212d2afe4476> CONTENT-LENGTH: 313 SUPPORTED: 100rel USER-AGENT: RTCC/3.5.0.0 MediationServer CONTENT-TYPE: application/sdp; charset=utf-8 ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite v=0 o=- 120 1 IN IP4 1.1.7.7 s=session c=IN IP4 1.1.7.7 b=CT:1000 t=0 0 m=audio 63818 RTP/AVP 97 101 13 0 8 c=IN IP4 1.1.7.7 a=rtcp:63819 a=label:Audio a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 [5] 2010/01/12 14:32:00: Identify trunk (IP address and domain match) 2 [7] 2010/01/12 14:32:00: SIP Tx tcp:1.1.7.7:58486: SIP/2.0 100 Trying Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1 From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 CSeq: 113 INVITE Content-Length: 0 [7] 2010/01/12 14:32:00: Set packet length to 20 [6] 2010/01/12 14:32:00: Sending RTP for a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b to 1.1.7.7:63818 [5] 2010/01/12 14:32:00: Dialplan dialplan: Match +78632370684@1.1.1.5 to <sip:78632370684@sipnet.ru;user=phone> on trunk skype [5] 2010/01/12 14:32:00: Using <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 as redirect source address [7] 2010/01/12 14:32:00: SIP Tx udp:212.53.40.40:5060: INVITE sip:78632370684@sipnet.ru;user=phone SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone> Call-ID: b92b6ef2@pbx CSeq: 10060 INVITE Max-Forwards: 70 Contact: <sip:0025879052@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Related-Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 Content-Type: application/sdp Content-Length: 323 v=0 o=- 131 131 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=audio 59854 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2010/01/12 14:32:00: Set packet length to 20 [6] 2010/01/12 14:32:00: Send codec pcmu/8000 [7] 2010/01/12 14:32:00: SIP Tx tcp:1.1.7.7:58486: SIP/2.0 183 Ringing Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1 From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 CSeq: 113 INVITE Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 216 v=0 o=- 7916 7916 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=audio 50440 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2010/01/12 14:32:00: SIP Rx udp:212.53.40.40:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport=58937;received=212.176.115.249 From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone> Call-ID: b92b6ef2@pbx CSeq: 10060 INVITE Server: CommuniGatePro/5.2.19 Content-Length: 0 [7] 2010/01/12 14:32:00: SIP Rx udp:212.53.40.40:5060: SIP/2.0 401 Authentication required Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport=58937;received=212.176.115.249 From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone>;tag=7A36DD5F Call-ID: b92b6ef2@pbx CSeq: 10060 INVITE WWW-Authenticate: Digest realm="etc.tario.ru",nonce="740E87CA2557EC46E7DB",opaque="opaqueData",qop="auth",algorithm=MD5 Server: CommuniGatePro/5.2.19 Content-Length: 0 [7] 2010/01/12 14:32:00: SIP Tx udp:212.53.40.40:5060: ACK sip:78632370684@sipnet.ru;user=phone SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone>;tag=7A36DD5F Call-ID: b92b6ef2@pbx CSeq: 10060 ACK Max-Forwards: 70 Content-Length: 0 [7] 2010/01/12 14:32:00: SIP Tx udp:212.53.40.40:5060: INVITE sip:78632370684@sipnet.ru;user=phone SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-62d178b026c98eea459c2f4874670522;rport From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone> Call-ID: b92b6ef2@pbx CSeq: 10061 INVITE Max-Forwards: 70 Contact: <sip:0025879052@1.1.1.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Related-Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 Authorization: Digest realm="etc.tario.ru",nonce="740E87CA2557EC46E7DB",response="7c0198de271c502a6c46b23a7fb88568",username="0025879052",uri="sip:78632370684@sipnet.ru;user=phone",qop=auth,nc=00000001,cnonce="8dbfdfb6",opaque="opaqueData",algorithm=MD5 Content-Type: application/sdp Content-Length: 323 v=0 o=- 131 131 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=audio 59854 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 2010/01/12 14:32:00: SIP Rx udp:212.53.40.40:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-62d178b026c98eea459c2f4874670522;rport=58937;received=212.176.115.249 From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone> Call-ID: b92b6ef2@pbx CSeq: 10061 INVITE Server: CommuniGatePro/5.2.19 Content-Length: 0 [7] 2010/01/12 14:32:00: SIP Rx tcp:1.1.7.7:58486: PRACK sip:Anonymous@1.1.1.5:5060;transport=tcp SIP/2.0 FROM: <sip:+78632688634100@s-case1c.case.ru;user=phone>;epid=77B47EEE06;tag=a6cc0a75 TO: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b CSEQ: 114 PRACK CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK9fb71a0 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.5.0.0 MediationServer RAck: 1 113 INVITE [7] 2010/01/12 14:32:00: SIP Tx tcp:1.1.7.7:58486: SIP/2.0 200 Ok Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK9fb71a0 From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 CSeq: 114 PRACK Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp> User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Length: 0 [7] 2010/01/12 14:32:01: SIP Rx udp:212.53.40.40:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.1.1.5:5060;received=212.176.115.249;rport=58937;branch=z9hG4bK-62d178b026c98eea459c2f4874670522 Record-Route: <sip:212.53.35.244:5060;lr>,<sip:197897-192.168.40.71.dialog.cgatepro;lr> Record-Route: <sip:192.168.40.71:5060;lr> Record-Route: <sip:212.53.40.40:5060;lr> From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988 Call-ID: b92b6ef2@pbx CSeq: 10061 INVITE Content-Type: application/sdp Server: TarioSoftswitch/3.2.11 Content-Length: 231 v=0 o=Tario-Softswitch 2749 101 IN IP4 212.53.40.91 s=SIP Call c=IN IP4 212.53.40.71 t=0 0 m=audio 26520 RTP/AVP 8 97 c=IN IP4 212.53.40.71 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=ptime:20 [7] 2010/01/12 14:32:01: Set packet length to 20 [6] 2010/01/12 14:32:01: Send codec=pcma/8000 afrer answer [6] 2010/01/12 14:32:01: Sending RTP for b92b6ef2@pbx#55270 to 212.53.40.71:26520 [7] 2010/01/12 14:32:01: b92b6ef2@pbx#55270: RTP pass-through mode [7] 2010/01/12 14:32:01: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: RTP pass-through mode [7] 2010/01/12 14:32:01: Cannot pass through on b92b6ef2@pbx#55270, falling back to transcoding [7] 2010/01/12 14:32:04: SIP Rx udp:212.53.40.40:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.5:5060;received=212.176.115.249;rport=58937;branch=z9hG4bK-62d178b026c98eea459c2f4874670522 Record-Route: <sip:212.53.35.244:5060;lr>,<sip:197897-192.168.40.71.dialog.cgatepro;lr> Record-Route: <sip:192.168.40.71:5060;lr> Record-Route: <sip:212.53.40.40:5060;lr> From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988 Call-ID: b92b6ef2@pbx CSeq: 10061 INVITE Contact: <sip:proc-5282988@212.53.35.244> Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS Server: TarioSoftswitch/3.2.11 Content-Length: 231 v=0 o=Tario-Softswitch 2749 101 IN IP4 212.53.40.91 s=SIP Call c=IN IP4 212.53.40.71 t=0 0 m=audio 26520 RTP/AVP 8 97 c=IN IP4 212.53.40.71 a=rtpmap:8 PCMA/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=ptime:20 [7] 2010/01/12 14:32:04: Call b92b6ef2@pbx#55270: Clear last INVITE [7] 2010/01/12 14:32:04: Set packet length to 20 [7] 2010/01/12 14:32:04: SIP Tx udp:212.53.40.40:5060: ACK sip:proc-5282988@212.53.35.244 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-1d6e4f20cb81caac2bbcd1f8e52266dc;rport Route: <sip:212.53.40.40:5060;lr> Route: <sip:192.168.40.71:5060;lr> Route: <sip:197897-192.168.40.71.dialog.cgatepro;lr> Route: <sip:212.53.35.244:5060;lr> From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 To: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988 Call-ID: b92b6ef2@pbx CSeq: 10061 ACK Max-Forwards: 70 Contact: <sip:0025879052@1.1.1.5:5060;transport=udp> Authorization: Digest realm="etc.tario.ru",nonce="740E87CA2557EC46E7DB",response="aae322d85299e2b96309fcda4830667c",username="0025879052",uri="sip:proc-5282988@212.53.35.244",qop=auth,nc=00000002,cnonce="90704f9a",opaque="opaqueData",algorithm=MD5 Content-Length: 0 [7] 2010/01/12 14:32:04: Determine pass-through mode after receiving response [7] 2010/01/12 14:32:04: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: RTP pass-through mode [7] 2010/01/12 14:32:04: Cannot pass through on b92b6ef2@pbx#55270, falling back to transcoding [7] 2010/01/12 14:32:04: SIP Tx tcp:1.1.7.7:58486: SIP/2.0 200 Ok Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1 From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 CSeq: 113 INVITE Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 Content-Type: application/sdp Content-Length: 216 v=0 o=- 7916 7916 IN IP4 1.1.1.5 s=- c=IN IP4 1.1.1.5 t=0 0 m=audio 50440 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [7] 2010/01/12 14:32:04: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: RTP pass-through mode [7] 2010/01/12 14:32:04: SIP Rx tcp:1.1.7.7:58486: ACK sip:Anonymous@1.1.1.5:5060;transport=tcp SIP/2.0 FROM: <sip:+78632688634100@s-case1c.case.ru;user=phone>;epid=77B47EEE06;tag=a6cc0a75 TO: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b CSEQ: 113 ACK CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK7e1d4647 CONTENT-LENGTH: 0 USER-AGENT: RTCC/3.5.0.0 MediationServer [7] 2010/01/12 14:32:04: Cannot pass through on b92b6ef2@pbx#55270, falling back to transcoding [7] 2010/01/12 14:32:07: SIP Rx udp:1.1.5.149:5060: SUBSCRIBE sip:1.1.1.5:53242;transport=tcp SIP/2.0 From: <sip:643@1.1.1.5>;tag=1010595-13c4-0-31b-2443 To: <sip:643@1.1.1.5>;tag=3f18c3a09b Call-ID: 806e43ec-1010595-13c4-0-316-54fd@1.1.1.5 CSeq: 631 SUBSCRIBE Via: SIP/2.0/UDP 1.1.5.149:5060;branch=z9hG4bK-763a-1cdd4c0-234 Expires: 47 Event: message-summary Max-Forwards: 70 Supported: replaces,100rel Accept: application/simple-message-summary Contact: <sip:643@1.1.5.149:5060;transport=TCP> Content-Length: 0 [7] 2010/01/12 14:32:07: SIP Tx udp:1.1.5.149:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 1.1.5.149:5060;branch=z9hG4bK-763a-1cdd4c0-234 From: <sip:643@1.1.1.5>;tag=1010595-13c4-0-31b-2443 To: <sip:643@1.1.1.5>;tag=3f18c3a09b Call-ID: 806e43ec-1010595-13c4-0-316-54fd@1.1.1.5 CSeq: 631 SUBSCRIBE Contact: <sip:1.1.1.5:5060;transport=udp> Expires: 48 Content-Length: 0 [7] 2010/01/12 14:32:09: Cannot pass through on a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b, falling back to transcoding [7] 2010/01/12 14:32:09: Received RFC4733 DTMF on codec 101 [5] 2010/01/12 14:32:10: Tuning to new SSRC [5] 2010/01/12 14:32:14: Last message repeated 2 times [7] 2010/01/12 14:32:14: SIP Rx udp:212.53.40.40:5060: BYE sip:0025879052@1.1.1.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK480354-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK263354062 Via: SIP/2.0/TCP 212.53.35.244:50666;branch=z9hG4bK-2c5d5000-5282988 Max-Forwards: 68 From: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988 To: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 Call-ID: b92b6ef2@pbx CSeq: 35426 BYE User-Agent: TarioSoftswitch/3.2.11 Content-Length: 0 [7] 2010/01/12 14:32:14: SIP Tx udp:212.53.40.40:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK480354-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport=5060 Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK263354062 Via: SIP/2.0/TCP 212.53.35.244:50666;branch=z9hG4bK-2c5d5000-5282988 From: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988 To: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270 Call-ID: b92b6ef2@pbx CSeq: 35426 BYE Contact: <sip:0025879052@1.1.1.5:5060;transport=udp> User-Agent: pbxnsip-PBX/3.4.0.3201 RTP-RxStat: Dur=14,Pkt=723,Oct=109636,Underun=458 RTP-TxStat: Dur=10,Pkt=41,Oct=1904 Content-Length: 0 [7] 2010/01/12 14:32:14: Other Ports: 1 [7] 2010/01/12 14:32:14: Call Port: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b [7] 2010/01/12 14:32:14: SIP Tx tcp:1.1.7.7:58486: BYE sip:s-case1c.case.ru:5060;transport=Tcp;maddr=1.1.7.7;ms-opaque=d799212d2afe4476 SIP/2.0 Via: SIP/2.0/TCP 1.1.1.5:5060;branch=z9hG4bK-06141a693318556b7acae6c0d7f66226;rport From: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b To: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 CSeq: 9721 BYE Max-Forwards: 70 Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp> RTP-RxStat: Dur=14,Pkt=39,Oct=1092,Underun=2 RTP-TxStat: Dur=10,Pkt=678,Oct=116616 Content-Length: 0 [7] 2010/01/12 14:32:14: SIP Rx tcp:1.1.7.7:58486: SIP/2.0 200 OK FROM: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b TO: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 CSEQ: 9721 BYE CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956 VIA: SIP/2.0/TCP 1.1.1.5:5060;branch=z9hG4bK-06141a693318556b7acae6c0d7f66226;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.5.0.0 MediationServer [7] 2010/01/12 14:32:14: Call a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: Clear last request [5] 2010/01/12 14:32:14: BYE Response: Terminate a6546ac6-7af5-4a17-8856-0f66a5817956
  11. ShadowAnt
    I did it! no problem to connect PBXnSIP to sip provider like Skype for SIP or our Russian sipnet.ru
  12. ShadowAnt
    Ok, Thank you!
  13. ShadowAnt

    REFER

    ShadowAnt replied to hosted's topic in Microsoft Exchange

    Thank you! If I understand, I must wait for some changes in PBXnSIP for Exchange 2010 fax transfer? or I can use another FoIP solution...
  14. ShadowAnt
    Hello, dear friends! some time ago, Skype is started beta "Skype for SIP". I have login, password and Skype SIP server address... How can I connect my PBXnSIP to this service? Maybe anyone have any practise?
  15. ShadowAnt

    REFER

    ShadowAnt replied to hosted's topic in Microsoft Exchange

    hello! any ideas?
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