История изменений [назад]

не могу понять почему не ходят факсы. История такая, asterisk 1.8.8.0 spandsp 0.0.6pre18 core show capabilities Registered FAX Technology Modules: Type : Spandsp Description : Spandsp FAX Driver Capabilities : SEND RECEIVE T.38 G.711

В sip show settings T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1

а вот дебаг`Content-Type: application/sdp Content-Length: 254

v=0 o=Essentra-Relay 4268132398 2 IN IP4 91.231.214.10 s=- c=IN IP4 91.231.214.10 t=0 0 m=image 39164 udptl t38 a=T38FaxUdpEC:t38UDPFEC a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:238 <-------------> --- (10 headers 11 lines) --- Got T.38 offer in SDP in dialog 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Transmitting (NAT) to 91.231.214.10:5060: ACK sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6bc57d08;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.8.0) Content-Length: 0

Really destroying SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 91.231.214.10:5060: REGISTER sip:voip.comtelco.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK61015e00;rport Max-Forwards: 70 From: <sip:74956986112@voip.comtelco.ru>;tag=as2127f6b8 To: <sip:74956986112@voip.comtelco.ru> Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 106 REGISTER User-Agent: FPBX-2.9.0(1.8.8.0) Authorization: Digest username="74956986112", realm="74956986112", algorithm=MD5, uri="sip:voip.comtelco.ru", nonce="7275753bc67b3108843853f03e4c48ad", response="edba6bfcd920dda04c4b8168003a3540", qop=auth, cnonce="4dd7c338", nc=00000004 Expires: 120 Contact: <sip:74956986112@192.168.0.4:5060> Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@voip.comtelco.ru>;tag=as2127f6b8 To: <sip:74956986112@voip.comtelco.ru>;tag=e2adfb0 Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 106 REGISTER Expires: 30 Server: vocl-essentra-bax/8.0.251 Contact: <sip:74956986112@192.168.0.4:5060> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK61015e00 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' in 32000 ms (Method: REGISTER) setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 91.231.214.10:5060: INVITE sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK4e28dedd;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.8.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 282

v=0 o=root 1496357654 1496357656 IN IP4 192.168.0.4 s=Asterisk PBX 1.8.8.0 c=IN IP4 192.168.0.4 t=0 0 m=audio 10428 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 100 Trying From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=1536fb08 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK4e28dedd Content-Length: 0

<-------------> --- (7 headers 0 lines) ---

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE Server: vocl-essentra-bax/8.0.251 Contact: <sip:+12314987446@91.231.214.10;vtservice=callcontrol.callcontrolservlet> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK4e28dedd Content-Type: application/sdp Content-Length: 222

v=0 o=Essentra-Relay 4268132398 3 IN IP4 91.231.214.10 s=- c=IN IP4 91.231.214.10 t=0 0 m=audio 39164 RTP/AVP 0 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 96 Found audio description format telephone-event for ID 96 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 91.231.214.10:39164 setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Transmitting (NAT) to 91.231.214.10:5060: ACK sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5f673cea;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.8.0) Content-Length: 0

-- Executing [s@ext-fax:5] ExecIf("SIP/outComtelko-00000003", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack
-- Executing [s@ext-fax:6] Hangup("SIP/outComtelko-00000003", "") in new stack

== Spawn extension (ext-fax, s, 6) exited non-zero on 'SIP/outComtelko-00000003' -- Executing [h@ext-fax:1] GotoIf("SIP/outComtelko-00000003", "1?failed") in new stack -- Goto (ext-fax,h,103) -- Executing [h@ext-fax:103] NoOp("SIP/outComtelko-00000003", "FAX "FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries" for: artec@agava.com , From: "12314987446" <+12314987446>") in new stack -- Executing [h@ext-fax:104] Macro("SIP/outComtelko-00000003", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/outComtelko-00000003", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] Hangup("SIP/outComtelko-00000003", "") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/outComtelko-00000003' in macro 'hangupcall' == Spawn extension (ext-fax, h, 104) exited non-zero on 'SIP/outComtelko-00000003' Scheduling destruction of SIP dialog '11ED98B0-05EF-4E8E-97C4-7646CBABE7A0' in 32000 ms (Method: ACK) setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Reliably Transmitting (NAT) to 91.231.214.10:5060: BYE sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK422de7be;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 104 BYE User-Agent: FPBX-2.9.0(1.8.8.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 104 BYE Server: vocl-essentra-bax/8.0.251 Contact: <sip:+12314987446@91.231.214.10;vtservice=callcontrol.callcontrolservlet> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK422de7be Content-Length: 0

<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '11ED98B0-05EF-4E8E-97C4-7646CBABE7A0' Method: ACK Really destroying SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 91.231.214.10:5060: REGISTER sip:voip.comtelco.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6bb9b09f;rport Max-Forwards: 70 From: <sip:74956986112@voip.comtelco.ru>;tag=as3d613399 To: <sip:74956986112@voip.comtelco.ru> Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 107 REGISTER User-Agent: FPBX-2.9.0(1.8.8.0) Authorization: Digest username="74956986112", realm="74956986112", algorithm=MD5, uri="sip:voip.comtelco.ru", nonce="7275753bc67b3108843853f03e4c48ad", response="a607a0a934401adcd2162829291db58c", qop=auth, cnonce="36481b04", nc=00000005 Expires: 120 Contact: <sip:74956986112@192.168.0.4:5060> Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@voip.comtelco.ru>;tag=as3d613399 To: <sip:74956986112@voip.comtelco.ru>;tag=13176358 Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 107 REGISTER Expires: 30 Server: vocl-essentra-bax/8.0.251 Contact: <sip:74956986112@192.168.0.4:5060> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK6bb9b09f Content-Length: 0 `

не могу понять почему не ходят факсы. История такая, asterisk 1.8.8.0 spandsp 0.0.6pre18 core show capabilities Registered FAX Technology Modules: Type : Spandsp Description : Spandsp FAX Driver Capabilities : SEND RECEIVE T.38 G.711

В sip show settings T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1

а вот дебаг`Content-Type: дебаг

`Content-Type: application/sdp

Content-Length: 254

v=0 o=Essentra-Relay 4268132398 2 IN IP4 91.231.214.10 s=- c=IN IP4 91.231.214.10 t=0 0 m=image 39164 udptl t38 a=T38FaxUdpEC:t38UDPFEC a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:238 <-------------> --- (10 headers 11 lines) --- Got T.38 offer in SDP in dialog 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Transmitting (NAT) to 91.231.214.10:5060: ACK sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6bc57d08;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.8.0) Content-Length: 0

Really destroying SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 91.231.214.10:5060: REGISTER sip:voip.comtelco.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK61015e00;rport Max-Forwards: 70 From: <sip:74956986112@voip.comtelco.ru>;tag=as2127f6b8 To: <sip:74956986112@voip.comtelco.ru> Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 106 REGISTER User-Agent: FPBX-2.9.0(1.8.8.0) Authorization: Digest username="74956986112", realm="74956986112", algorithm=MD5, uri="sip:voip.comtelco.ru", nonce="7275753bc67b3108843853f03e4c48ad", response="edba6bfcd920dda04c4b8168003a3540", qop=auth, cnonce="4dd7c338", nc=00000004 Expires: 120 Contact: <sip:74956986112@192.168.0.4:5060> Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@voip.comtelco.ru>;tag=as2127f6b8 To: <sip:74956986112@voip.comtelco.ru>;tag=e2adfb0 Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 106 REGISTER Expires: 30 Server: vocl-essentra-bax/8.0.251 Contact: <sip:74956986112@192.168.0.4:5060> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK61015e00 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' in 32000 ms (Method: REGISTER) setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 91.231.214.10:5060: INVITE sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK4e28dedd;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.8.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 282

v=0 o=root 1496357654 1496357656 IN IP4 192.168.0.4 s=Asterisk PBX 1.8.8.0 c=IN IP4 192.168.0.4 t=0 0 m=audio 10428 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 100 Trying From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=1536fb08 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK4e28dedd Content-Length: 0

<-------------> --- (7 headers 0 lines) ---

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE Server: vocl-essentra-bax/8.0.251 Contact: <sip:+12314987446@91.231.214.10;vtservice=callcontrol.callcontrolservlet> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK4e28dedd Content-Type: application/sdp Content-Length: 222

v=0 o=Essentra-Relay 4268132398 3 IN IP4 91.231.214.10 s=- c=IN IP4 91.231.214.10 t=0 0 m=audio 39164 RTP/AVP 0 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 96 Found audio description format telephone-event for ID 96 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 91.231.214.10:39164 setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Transmitting (NAT) to 91.231.214.10:5060: ACK sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5f673cea;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.8.0) Content-Length: 0

-- Executing [s@ext-fax:5] ExecIf("SIP/outComtelko-00000003", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack
-- Executing [s@ext-fax:6] Hangup("SIP/outComtelko-00000003", "") in new stack

== Spawn extension (ext-fax, s, 6) exited non-zero on 'SIP/outComtelko-00000003' -- Executing [h@ext-fax:1] GotoIf("SIP/outComtelko-00000003", "1?failed") in new stack -- Goto (ext-fax,h,103) -- Executing [h@ext-fax:103] NoOp("SIP/outComtelko-00000003", "FAX "FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries" for: artec@agava.com , From: "12314987446" <+12314987446>") in new stack -- Executing [h@ext-fax:104] Macro("SIP/outComtelko-00000003", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/outComtelko-00000003", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] Hangup("SIP/outComtelko-00000003", "") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/outComtelko-00000003' in macro 'hangupcall' == Spawn extension (ext-fax, h, 104) exited non-zero on 'SIP/outComtelko-00000003' Scheduling destruction of SIP dialog '11ED98B0-05EF-4E8E-97C4-7646CBABE7A0' in 32000 ms (Method: ACK) setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Reliably Transmitting (NAT) to 91.231.214.10:5060: BYE sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK422de7be;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 104 BYE User-Agent: FPBX-2.9.0(1.8.8.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 104 BYE Server: vocl-essentra-bax/8.0.251 Contact: <sip:+12314987446@91.231.214.10;vtservice=callcontrol.callcontrolservlet> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK422de7be Content-Length: 0

<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '11ED98B0-05EF-4E8E-97C4-7646CBABE7A0' Method: ACK Really destroying SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 91.231.214.10:5060: REGISTER sip:voip.comtelco.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6bb9b09f;rport Max-Forwards: 70 From: <sip:74956986112@voip.comtelco.ru>;tag=as3d613399 To: <sip:74956986112@voip.comtelco.ru> Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 107 REGISTER User-Agent: FPBX-2.9.0(1.8.8.0) Authorization: Digest username="74956986112", realm="74956986112", algorithm=MD5, uri="sip:voip.comtelco.ru", nonce="7275753bc67b3108843853f03e4c48ad", response="a607a0a934401adcd2162829291db58c", qop=auth, cnonce="36481b04", nc=00000005 Expires: 120 Contact: <sip:74956986112@192.168.0.4:5060> Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@voip.comtelco.ru>;tag=as3d613399 To: <sip:74956986112@voip.comtelco.ru>;tag=13176358 Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 107 REGISTER Expires: 30 Server: vocl-essentra-bax/8.0.251 Contact: <sip:74956986112@192.168.0.4:5060> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK6bb9b09f Content-Length: 0 `0

не могу понять почему не ходят факсы. История такая, asterisk 1.8.8.0 spandsp 0.0.6pre18 core show capabilities Registered FAX Technology Modules: Type : Spandsp Description : Spandsp FAX Driver Capabilities : SEND RECEIVE T.38 G.711

В sip show settings T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1

а вот дебаг

`Content-Type: дебаг
` Content-Type: application/sdp

Content-Length: 254

v=0 o=Essentra-Relay 4268132398 2 IN IP4 91.231.214.10 s=- c=IN IP4 91.231.214.10 t=0 0 m=image 39164 udptl t38 a=T38FaxUdpEC:t38UDPFEC a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:238 <-------------> --- (10 headers 11 lines) --- Got T.38 offer in SDP in dialog 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Transmitting (NAT) to 91.231.214.10:5060: ACK sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6bc57d08;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 102 ACK User-Agent: FPBX-2.9.0(1.8.8.0) Content-Length: 0

Really destroying SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 91.231.214.10:5060: REGISTER sip:voip.comtelco.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK61015e00;rport Max-Forwards: 70 From: <sip:74956986112@voip.comtelco.ru>;tag=as2127f6b8 To: <sip:74956986112@voip.comtelco.ru> Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 106 REGISTER User-Agent: FPBX-2.9.0(1.8.8.0) Authorization: Digest username="74956986112", realm="74956986112", algorithm=MD5, uri="sip:voip.comtelco.ru", nonce="7275753bc67b3108843853f03e4c48ad", response="edba6bfcd920dda04c4b8168003a3540", qop=auth, cnonce="4dd7c338", nc=00000004 Expires: 120 Contact: <sip:74956986112@192.168.0.4:5060> Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@voip.comtelco.ru>;tag=as2127f6b8 To: <sip:74956986112@voip.comtelco.ru>;tag=e2adfb0 Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 106 REGISTER Expires: 30 Server: vocl-essentra-bax/8.0.251 Contact: <sip:74956986112@192.168.0.4:5060> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK61015e00 Content-Length: 0

<-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' in 32000 ms (Method: REGISTER) setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 91.231.214.10:5060: INVITE sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK4e28dedd;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE User-Agent: FPBX-2.9.0(1.8.8.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 282

v=0 o=root 1496357654 1496357656 IN IP4 192.168.0.4 s=Asterisk PBX 1.8.8.0 c=IN IP4 192.168.0.4 t=0 0 m=audio 10428 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 100 Trying From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=1536fb08 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK4e28dedd Content-Length: 0

<-------------> --- (7 headers 0 lines) ---

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 INVITE Server: vocl-essentra-bax/8.0.251 Contact: <sip:+12314987446@91.231.214.10;vtservice=callcontrol.callcontrolservlet> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK4e28dedd Content-Type: application/sdp Content-Length: 222

v=0 o=Essentra-Relay 4268132398 3 IN IP4 91.231.214.10 s=- c=IN IP4 91.231.214.10 t=0 0 m=audio 39164 RTP/AVP 0 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (10 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 96 Found audio description format telephone-event for ID 96 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 91.231.214.10:39164 setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Transmitting (NAT) to 91.231.214.10:5060: ACK sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK5f673cea;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Contact: <sip:74956986112@192.168.0.4:5060> Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 103 ACK User-Agent: FPBX-2.9.0(1.8.8.0) Content-Length: 0

-- Executing [s@ext-fax:5] ExecIf("SIP/outComtelko-00000003", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack
-- Executing [s@ext-fax:6] Hangup("SIP/outComtelko-00000003", "") in new stack

== Spawn extension (ext-fax, s, 6) exited non-zero on 'SIP/outComtelko-00000003' -- Executing [h@ext-fax:1] GotoIf("SIP/outComtelko-00000003", "1?failed") in new stack -- Goto (ext-fax,h,103) -- Executing [h@ext-fax:103] NoOp("SIP/outComtelko-00000003", "FAX "FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries" for: artec@agava.com , From: "12314987446" <+12314987446>") in new stack -- Executing [h@ext-fax:104] Macro("SIP/outComtelko-00000003", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/outComtelko-00000003", "1?theend") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] Hangup("SIP/outComtelko-00000003", "") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/outComtelko-00000003' in macro 'hangupcall' == Spawn extension (ext-fax, h, 104) exited non-zero on 'SIP/outComtelko-00000003' Scheduling destruction of SIP dialog '11ED98B0-05EF-4E8E-97C4-7646CBABE7A0' in 32000 ms (Method: ACK) setdestination: Parsing <sip:91.231.214.10:5060> for address/port to send to setdestination: set destination to 91.231.214.10:5060 Reliably Transmitting (NAT) to 91.231.214.10:5060: BYE sip:91.231.214.10:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK422de7be;rport Max-Forwards: 70 From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 104 BYE User-Agent: FPBX-2.9.0(1.8.8.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@89.108.96.17:62272>;tag=as2f95a292 To: "12314987446"<sip:+12314987446@91.231.214.10:5060>;tag=21838 Call-ID: 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0 CSeq: 104 BYE Server: vocl-essentra-bax/8.0.251 Contact: <sip:+12314987446@91.231.214.10;vtservice=callcontrol.callcontrolservlet> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK422de7be Content-Length: 0

<-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '11ED98B0-05EF-4E8E-97C4-7646CBABE7A0' Method: ACK Really destroying SIP dialog '471cb7e344a7452958f01fec267c9eaf@0.0.0.1' Method: REGISTER REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 91.231.214.10:5060: REGISTER sip:voip.comtelco.ru SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK6bb9b09f;rport Max-Forwards: 70 From: <sip:74956986112@voip.comtelco.ru>;tag=as3d613399 To: <sip:74956986112@voip.comtelco.ru> Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 107 REGISTER User-Agent: FPBX-2.9.0(1.8.8.0) Authorization: Digest username="74956986112", realm="74956986112", algorithm=MD5, uri="sip:voip.comtelco.ru", nonce="7275753bc67b3108843853f03e4c48ad", response="a607a0a934401adcd2162829291db58c", qop=auth, cnonce="36481b04", nc=00000005 Expires: 120 Contact: <sip:74956986112@192.168.0.4:5060> Content-Length: 0

<--- SIP read from UDP:91.231.214.10:5060 ---> SIP/2.0 200 OK From: <sip:74956986112@voip.comtelco.ru>;tag=as3d613399 To: <sip:74956986112@voip.comtelco.ru>;tag=13176358 Call-ID: 471cb7e344a7452958f01fec267c9eaf@0.0.0.1 CSeq: 107 REGISTER Expires: 30 Server: vocl-essentra-bax/8.0.251 Contact: <sip:74956986112@192.168.0.4:5060> Via: SIP/2.0/UDP 192.168.0.4:5060;received=89.108.96.17;rport=62272;branch=z9hG4bK6bb9b09f Content-Length: 00 `

не могу понять почему не ходят факсы. История такая, asterisk 1.8.8.0 spandsp 0.0.6pre18 core show capabilities Registered FAX Technology Modules: Type : Spandsp Description : Spandsp FAX Driver Capabilities : SEND RECEIVE T.38 G.711

В sip show settings T.38 support: Yes T.38 EC mode: FEC T.38 MaxDtgrm: -1

-- Executing [s@ext-fax:5] ExecIf("SIP/outComtelko-00000003", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack
-- Executing [s@ext-fax:6] Hangup("SIP/outComtelko-00000003", "") in new stack

не могу понять почему не ходят факсы. История такая, asterisk 1.8.8.0 spandsp 0.0.6pre18 core show capabilities Registered FAX Technology Modules: Type : Spandsp Description : Spandsp FAX Driver Capabilities : SEND RECEIVE T.38 G.711

В sip show settings settings

T.38 support: Yes Yes

T.38 EC mode: FEC FEC

T.38 MaxDtgrm: -1

Content-Type: application/sdp
Content-Length: 254
v=0
o=Essentra-Relay 4268132398 2 IN IP4 91.231.214.10
s=-
c=IN IP4 91.231.214.10
t=0 0
m=image 39164 udptl t38
a=T38FaxUdpEC:t38UDPFEC
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
--- (10 headers 11 lines) ---
Got T.38 offer in SDP in dialog 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0          
nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing),        
combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

не могу понять почему не ходят факсы. История такая, asterisk 1.8.8.0 spandsp 0.0.6pre18 core show capabilities Registered FAX Technology Modules: Type : Spandsp Description : Spandsp FAX Driver Capabilities : SEND RECEIVE T.38 G.711

В sip show settings

T.38 support: Yes

T.38 EC mode: FEC

T.38 MaxDtgrm: -1

В sip.conf ничего не прописывал у меня freepbx freepbx

в sipgeneralcustom.conf sip_general_custom.conf

t38pt_udptl=yes

настройка транка username=7495XXXXXXX type=peer транка

username=7495XXXXXXX

type=peer

secret=* insecure=invite,port host=voip.comtelco.ru fromuser=7495XXXXX

insecure=invite,port

host=voip.comtelco.ru

fromuser=7495XXXXX

canreinvite=yes

Content-Type: application/sdp
Content-Length: 254
v=0
o=Essentra-Relay 4268132398 2 IN IP4 91.231.214.10
s=-
c=IN IP4 91.231.214.10
t=0 0
m=image 39164 udptl t38
a=T38FaxUdpEC:t38UDPFEC
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238
--- (10 headers 11 lines) ---
Got T.38 offer in SDP in dialog 11ED98B0-05EF-4E8E-97C4-7646CBABE7A0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0          
nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing),        
combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.