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ShadowAnt

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  1. I try it on 32-bit edition of windows server 2003, windows XP, but still have the same problem...
  2. The host have a static IP. maybe it is because windows 2008 x64...?
  3. <port_bind4/><port_bind6/> I didn't see any parameters in this string...
  4. 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. 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. 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. 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. 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. 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. 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. I did it! no problem to connect PBXnSIP to sip provider like Skype for SIP or our Russian sipnet.ru
  12. ShadowAnt

    REFER

    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...
  13. 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?
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