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GS-1004B исходящий звонок http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=2804	Страница 1 из 1

Автор:	Sergius [ 14 май 2012, 13:22 ]
Заголовок сообщения:	GS-1004B исходящий звонок
День добрый!!! Может кто подскажет ответ на мой вопрос, в инете внятного ответа не нашел... Ситуация такова: Имеется GS-1004B подцепленый к SIP-серверу (3CX). Все работает замечательно, за исключением одного момента - при исходящем звонке через GSM (FXO не используется), если вызываемый абонент поднял трубку и после разговора положил трубку, то отбой проходит нормально ( SIP/2.0 486). Если же вызываемый абонент не поднимая трубки делает отбой вызова, то шлюз присылает SIP/2.0 480, SIP-сервер думает что канал не доступен, и начинает звонить по резервным маршрутам (в моем случае перебирает все 4 СИМ-карты и еще уходит на резервный городской номер). Как его заставить передавать отбой в виде SIP/2.0 486?

Автор:	Denis [ 14 май 2012, 13:43 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
Можете дебаги выложить одновременно в двух случая? (абонент берет трубку и просто отбивает звонок) deb voip sip deb voip call deb rta ipc deb gsm 0 0 call deb gsm 0 0 rx deb gsm 0 0 mon

Автор:	Sergius [ 17 май 2012, 10:47 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
вот debug voip sip Вызывающий номер 7102 Вызываемый сотовый номер 7102 10.202.4.1 - SIP-сервер 10.202.4.3 - Addpac 1. Когда абонент отоветил и потом положил трубку. (нормально) Received SIP PDU from ( 10.202.4.1:5060 ) INVITE sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060> From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 421 v=0 o=3cxPS 173140869120 249175212033 IN IP4 10.202.4.1 s=3cxPS Audio call c=IN IP4 10.202.4.1 t=0 0 m=audio 7312 RTP/AVP 0 8 3 13 9 18 110 99 101 c=IN IP4 10.202.4.1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 iLBC/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 To: <sip:7102@10.202.4.3:5060> Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 To: <sip:7102@10.202.4.3:5060>;tag=0e4f1e1fa4 Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337265422 1337265422 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337265422 0 m=audio 23056 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-ef20fb70394c0b46-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 To: <sip:7102@10.202.4.3:5060>;tag=0e4f1e1fa4 Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337265432 1337265432 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337265432 0 m=audio 23056 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Received SIP PDU from ( 10.202.4.1:5060 ) ACK sip:7102@10.202.4.3 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-12218f51f9002765-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060>;tag=0e4f1e1fa4 From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 BYE sip:10013@10.202.4.1 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bK0e4f1e1fa45 From: <sip:7102@10.202.4.3:5060>;tag=0e4f1e1fa4 To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 5 BYE Date: Thu, 17 May 2012 14:37:14 GMT User-Agent: AddPac SIP Gateway Contact: <sip:7102@10.202.4.3> Content-Length: 0 Max-Forwards: 70 Received SIP PDU from ( 10.202.4.1:5060 ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bK0e4f1e1fa45 Contact: <sip:10013@10.202.4.1:5060> To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=4322c719 From: <sip:7102@10.202.4.3:5060>;tag=0e4f1e1fa4 Call-ID: MWY3NTQyNGZjZjNmY2M1MGU5ZWNkZThiOGJjY2E4YTk. CSeq: 5 BYE User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 0 2. Когда вызываемый абонент сразу отбил бызов. (не нормально) Received SIP PDU from ( 10.202.4.1:5060 ) INVITE sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060> From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925 Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 420 v=0 o=3cxPS 12448694272 290614935553 IN IP4 10.202.4.1 s=3cxPS Audio call c=IN IP4 10.202.4.1 t=0 0 m=audio 7278 RTP/AVP 0 8 3 13 9 18 110 99 101 c=IN IP4 10.202.4.1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 iLBC/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925 To: <sip:7102@10.202.4.3:5060> Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925 To: <sip:7102@10.202.4.3:5060>;tag=2f4f3b1ba4 Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337265199 1337265199 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337265199 0 m=audio 23048 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925 To: <sip:7102@10.202.4.3:5060>;tag=2f4f3b1ba4 Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Received SIP PDU from ( 10.202.4.1:5060 ) ACK sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-481430690248db03-1---d8754z-;rport Max-Forwards: 70 To: <sip:7102@10.202.4.3:5060>;tag=2f4f3b1ba4 From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=983b1925 Call-ID: MjRhYThkZGZmNGRkY2Q2ODQyZjNiNzM2OTYyYjAzYTk. CSeq: 1 ACK Content-Length: 0 Received SIP PDU from ( 10.202.4.1:5060 ) INVITE sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-d362384179604b46-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060> From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=7c20da3e Call-ID: MGM5MjliMGU4Mzc4MzIyZGUxYzBhYTA2NGM5NGZlOTQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 419 v=0 o=3cxPS 3657433088 297191604225 IN IP4 10.202.4.1 s=3cxPS Audio call c=IN IP4 10.202.4.1 t=0 0 m=audio 7280 RTP/AVP 0 8 3 13 9 18 110 99 101 c=IN IP4 10.202.4.1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 iLBC/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-d362384179604b46-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=7c20da3e To: <sip:7102@10.202.4.3:5060> Call-ID: MGM5MjliMGU4Mzc4MzIyZGUxYzBhYTA2NGM5NGZlOTQ. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-d362384179604b46-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=7c20da3e To: <sip:7102@10.202.4.3:5060>;tag=394fea1ca4 Call-ID: MGM5MjliMGU4Mzc4MzIyZGUxYzBhYTA2NGM5NGZlOTQ. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337265209 1337265209 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337265209 0 m=audio 23050 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv чуть позже другие дебаги сделаю...

Автор:	Sergius [ 17 май 2012, 11:03 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
вот debug voip call Выриант 1 (нормальный) 1 <Call 49> : **** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337266676) ** 2 <SIP 49> : Receive INVITE Request 3 <NetCon 49> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1) 4 <NetCon 49> : Found inbound voip peer by dest-pattern id(12201) 5 <NetCon 49> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3) 6 <Call 49> : From Net - calledParty(7102) callingParty(10013) 7 <Call 49> : MatchedPerfect 8 <Call 49> : MatchAllProcess After Sorted <0> id(4589) dest(....) prefer(0) selected(5) <1> id(4590) dest(....) prefer(0) selected(5) <2> id(4591) dest(....) prefer(0) selected(5) <3> id(4588) dest(....) prefer(0) selected(6) 9 <Call 49> : Initiate callee with dial-peer(....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 10 <CEP 010100> : InitiateOutCall : calledNum(7102), callingNum(10013), callerPort(ffffffff) type(GSM) 11 <CEP 010100> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(49) 12 <SIP 49> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 13 <SIP 49> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 14 <PhonePlay 49> : Audio Count(1) 15 <PhonePlay 49> : rtpSessionId(1) Second Audio Port(-1) 16 <SIP 49> : SetAlerting 17 <Call 49> : PreConnected from(10100) 18 <SIP 49> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 19 <SIP 49> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 20 <SIP 49> : Add Local Audio MediaFormat : 0 21 <Call 49> : Connected from(10100) 22 <SIP 49> : SetConnected 23 <SIP 49> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 24 <SIP 49> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 25 <SIP 49> : Add Local Audio MediaFormat : 0 26 <SIP 49> : ACK received 27 <SIP 49> : Receive ACK Request 28 <SIP 49> : Set Terminated Success for 1 INVITE 29 <CEP 010100> : Disconnected(16) at Busy 30 <Call 49> : Terminated from(10100) this(Local:CallClear) before((null)) forced(0) time(1337266688) 31 <SIP 49> : ReleaseWithBYE 32 <SIP 49> : Send BYE Request 33 <NetEP 49> : Call FROM <Sergey> terminated reason(Local:CallClear) 34 <CEP 010100> : DisconnectCall at Idle 35 <SIP 49> : Receive 200 OK 36 <SIP 49> : Transaction (6 BYE) completed Выриант 2(не нормальный) 1 <Call 47> : *** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337266569) ** 2 <SIP 47> : Receive INVITE Request 3 <NetCon 47> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1) 4 <NetCon 47> : Found inbound voip peer by dest-pattern id(12201) 5 <NetCon 47> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3) 6 <Call 47> : From Net - calledParty(7102) callingParty(10013) 7 <Call 47> : MatchedPerfect 8 <Call 47> : MatchAllProcess After Sorted <0> id(4591) dest(....) prefer(0) selected(4) <1> id(4588) dest(....) prefer(0) selected(5) <2> id(4589) dest(....) prefer(0) selected(5) <3> id(4590) dest(....) prefer(0) selected(5) 9 <Call 47> : Initiate callee with dial-peer(....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 10 <CEP 010300> : InitiateOutCall : calledNum(7102), callingNum(10013), callerPort(ffffffff) type(GSM) 11 <CEP 010300> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(47) 12 <SIP 47> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 13 <SIP 47> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 14 <PhonePlay 47> : Audio Count(1) 15 <PhonePlay 47> : rtpSessionId(1) Second Audio Port(-1) 16 <SIP 47> : SetAlerting 17 <Call 47> : PreConnected from(10300) 18 <SIP 47> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 19 <SIP 47> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 20 <SIP 47> : Add Local Audio MediaFormat : 0 21 <CEP 010300> : Disconnected(16) at Busy 22 <Call 47> : Terminated from(10300) this(Local:CallClear) before((null)) forced(0) time(1337266578) 23 <NetEP 47> : Call FROM <Sergey> terminated reason(Local:CallClear) 24 <CEP 010300> : DisconnectCall at Idle 25 <SIP 47> : Receive ACK Request 26 <Call 48> : *** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337266578) ** 27 <SIP 48> : Receive INVITE Request 28 <NetCon 48> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1) 29 <NetCon 48> : Found inbound voip peer by dest-pattern id(12201) 30 <NetCon 48> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3) 31 <Call 48> : From Net - calledParty(7102) callingParty(10013) 32 <Call 48> : MatchedPerfect 33 <Call 48> : MatchAllProcess After Sorted <0> id(4588) dest(....) prefer(0) selected(5) <1> id(4589) dest(....) prefer(0) selected(5) <2> id(4590) dest(....) prefer(0) selected(5) <3> id(4591) dest(....) prefer(0) selected(5) 34 <Call 48> : Initiate callee with dial-peer(....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 35 <CEP 010000> : InitiateOutCall : calledNum(*7102), callingNum(10013), callerPort(ffffffff) type(GSM) 36 <CEP 010000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(48) 37 <SIP 48> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 38 <SIP 48> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 39 <PhonePlay 48> : Audio Count(1) 40 <PhonePlay 48> : rtpSessionId(1) Second Audio Port(-1) 41 <SIP 48> : SetAlerting 42 <Call 48> : PreConnected from(10000) 43 <SIP 48> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 44 <SIP 48> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 45 <SIP 48> : Add Local Audio MediaFormat : 0

Автор:	Sergius [ 17 май 2012, 11:09 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
Вот debug rta ipc Вариант 1 (нормальный) [2891.255] RTA(1/0/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1) [2891.255] VP(1/0/0) open channel [2891.255] VM(1/0/0) Tx GSM CallRequest 1 stage 7102 [2891.255] VM(1/0/0) set T38 enable by CCC [2891.255] VM(1/0/0) set T38 mode STD [2891.255] VM(1/0/0) Fax rate 9600 [2891.255] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [2891.255] VM(1/0/0) VAD disable [2891.255] VP(1/0/0) update VAD 0 [2891.255] VM(1/0/0) SID enable by CCC [2891.270] RTA(1/0/0) Rx RS_OPEN_REQ callId=52 ssId=1 G711U peer=10.202.4.1 mp=23104/23105 hp=7050/7051 [2891.270] VM(1/0/0) codec same G711U [2891.270] RTA(1/0/0) Rx RS_LISTEN_REQ callId=52 ssId=1 G711U peer=10.202.4.1 mp=23104/23105 hp=7050/7051 [2899.055] RTA(1/0/0) Rx GSM_STTS_IND CALL_CONN [2899.055] RTA(1/0/0) Rx GSM_STTS_CALL_CONN at state=5 [2899.055] VP(1/0/0) attribute Fax enable, Modem disable [2899.055] VP(1/0/0) update Fax enable, Modem disable [2899.055] VM(1/0/0) Tx CONNECT_CNF [2899.055] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [2899.055] VM(1/0/0) VAD disable [2899.055] VM(1/0/0) SID enable by CCC [2899.070] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [2899.070] VM(1/0/0) DTMF_RTP_RFC2833 enable [2899.070] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65 [2901.780] RTA(1/0/0) Rx GSM_STTS_IND CALL_DISC [2901.780] RTA(1/0/0) Rx GSM_STTS_CALL_DISC at state=6 [2901.780] VM(1/0/0) vopp idle [2901.780] VP(1/0/0) close channel [2901.780] VM(1/0/0) Tx DISCONN_CNF [2901.790] RTA(1/0/0) Rx RS_CLOSE_REQ callId=52 ssId=1 dir=all [2901.790] RTA(1/0/0) close Media socket [2901.790] RTA(1/0/0) close RTCP socket Вариант 2 (не нормальный) [2810.065] RTA(1/2/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1) [2810.065] VP(1/2/0) open channel [2810.065] VM(1/2/0) Tx GSM CallRequest 1 stage 7102 [2810.065] VM(1/2/0) set T38 enable by CCC [2810.065] VM(1/2/0) set T38 mode STD [2810.065] VM(1/2/0) Fax rate 9600 [2810.065] RTA(1/2/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [2810.065] VM(1/2/0) VAD disable [2810.065] VP(1/2/0) update VAD 0 [2810.065] VM(1/2/0) SID enable by CCC [2810.080] RTA(1/2/0) Rx RS_OPEN_REQ callId=50 ssId=1 G711U peer=10.202.4.1 mp=23100/23101 hp=7040/7041 [2810.080] VM(1/2/0) codec same G711U [2810.080] RTA(1/2/0) Rx RS_LISTEN_REQ callId=50 ssId=1 G711U peer=10.202.4.1 mp=23100/23101 hp=7040/7041 [2818.665] RTA(1/2/0) Rx GSM_STTS_IND CALL_DISC [2818.665] RTA(1/2/0) Rx GSM_STTS_CALL_DISC at state=5 [2818.665] VM(1/2/0) vopp idle [2818.665] VP(1/2/0) close channel [2818.665] VM(1/2/0) Tx DISCONN_CNF [2818.665] RTA(1/2/0) Rx RS_CLOSE_REQ callId=50 ssId=1 dir=all [2818.665] RTA(1/2/0) close Media socket [2818.665] RTA(1/2/0) close RTCP socket [2818.795] RTA(1/3/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1) [2818.795] VP(1/3/0) open channel [2818.795] VM(1/3/0) Tx GSM CallRequest 1 stage *7102 [2818.795] VM(1/3/0) set T38 enable by CCC [2818.795] VM(1/3/0) set T38 mode STD [2818.795] VM(1/3/0) Fax rate 9600 [2818.800] RTA(1/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [2818.800] VM(1/3/0) VAD disable [2818.800] VP(1/3/0) update VAD 0 [2818.800] VM(1/3/0) SID enable by CCC [2818.810] RTA(1/3/0) Rx RS_OPEN_REQ callId=51 ssId=1 G711U peer=10.202.4.1 mp=23102/23103 hp=7042/7043 [2818.810] VM(1/3/0) codec same G711U [2818.810] RTA(1/3/0) Rx RS_LISTEN_REQ callId=51 ssId=1 G711U peer=10.202.4.1 mp=23102/23103 hp=7042/7043

Автор:	Sergius [ 17 май 2012, 11:15 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
deb gsm 1 0 rx Вариант 1 (нормальный) GSM-1/0-RSP: [0,0] 0D GSM-1/0-RSP: [0,0] 0A GSM-1/0-RSP: [0,0] 2B GSM-1/0-RSP: [0,1] 57 GSM-1/0-RSP: [0,2] 49 GSM-1/0-RSP: [0,3] 4E GSM-1/0-RSP: [0,4] 44 GSM-1/0-RSP: [0,5] 3A GSM-1/0-RSP: [0,6] 20 GSM-1/0-RSP: [0,7] 35 GSM-1/0-RSP: [0,8] 2C GSM-1/0-RSP: [0,9] 31 GSM-1/0-RSP: [0,10] 0D GSM-1/0-RSP: [0,10] 0A [3325.826] GSM-1/0: RECV, Line 0 : +WIND: 5,1 [3325.826] GSM-1/0: RECV, Line 0 : 2B 57 49 4E 44 3A 20 35 2C 31 GSM-1/0-RSP: [1,0] 0D GSM-1/0-RSP: [1,0] 0A GSM-1/0-RSP: [1,0] 2B GSM-1/0-RSP: [1,1] 57 GSM-1/0-RSP: [1,2] 49 GSM-1/0-RSP: [1,3] 4E GSM-1/0-RSP: [1,4] 44 GSM-1/0-RSP: [1,5] 3A GSM-1/0-RSP: [1,6] 20 GSM-1/0-RSP: [1,7] 32 GSM-1/0-RSP: [1,8] 0D GSM-1/0-RSP: [1,8] 0A [3332.971] GSM-1/0: RECV, Line 1 : +WIND: 2 [3332.971] GSM-1/0: RECV, Line 1 : 2B 57 49 4E 44 3A 20 32 GSM-1/0-RSP: [2,0] 0D GSM-1/0-RSP: [2,0] 0A GSM-1/0-RSP: [2,0] 4F GSM-1/0-RSP: [2,1] 4B GSM-1/0-RSP: [2,2] 0D GSM-1/0-RSP: [2,2] 0A [3334.380] GSM-1/0: RECV, Line 2 : OK [3334.380] GSM-1/0: RECV, Line 2 : 4F 4B GSM-1/0-EVT: [0] 0D GSM-1/0-EVT: [0] 0A GSM-1/0-EVT: [0] 4E GSM-1/0-EVT: [1] 4F GSM-1/0-EVT: [2] 20 GSM-1/0-EVT: [3] 43 GSM-1/0-EVT: [4] 41 GSM-1/0-EVT: [5] 52 GSM-1/0-EVT: [6] 52 GSM-1/0-EVT: [7] 49 GSM-1/0-EVT: [8] 45 GSM-1/0-EVT: [9] 52 GSM-1/0-EVT: [10] 0D GSM-1/0-EVT: [10] 0A GSM-1/0-EVT: [0] 0D GSM-1/0-EVT: [0] 0A GSM-1/0-EVT: [0] 2B GSM-1/0-EVT: [1] 57 GSM-1/0-EVT: [2] 49 GSM-1/0-EVT: [3] 4E GSM-1/0-EVT: [4] 44 GSM-1/0-EVT: [5] 3A GSM-1/0-EVT: [6] 20 GSM-1/0-EVT: [7] 36 GSM-1/0-EVT: [8] 2C GSM-1/0-EVT: [9] 31 GSM-1/0-EVT: [10] 0D GSM-1/0-EVT: [10] 0A GSM-1/0-RSP: [0,0] 0D GSM-1/0-RSP: [0,0] 0A GSM-1/0-RSP: [0,0] 2B GSM-1/0-RSP: [0,1] 43 GSM-1/0-RSP: [0,2] 53 GSM-1/0-RSP: [0,3] 51 GSM-1/0-RSP: [0,4] 3A GSM-1/0-RSP: [0,5] 20 GSM-1/0-RSP: [0,6] 31 GSM-1/0-RSP: [0,7] 35 GSM-1/0-RSP: [0,8] 2C GSM-1/0-RSP: [0,9] 30 GSM-1/0-RSP: [0,10] 0D GSM-1/0-RSP: [0,10] 0A [3339.772] GSM-1/0: RECV, Line 0 : +CSQ: 15,0 [3339.772] GSM-1/0: RECV, Line 0 : 2B 43 53 51 3A 20 31 35 2C 30 GSM-1/0-RSP: [1,0] 0D GSM-1/0-RSP: [1,0] 0A GSM-1/0-RSP: [1,0] 4F GSM-1/0-RSP: [1,1] 4B GSM-1/0-RSP: [1,2] 0D GSM-1/0-RSP: [1,2] 0A [3339.772] GSM-1/0: RECV, Line 1 : OK [3339.772] GSM-1/0: RECV, Line 1 : 4F 4B Вариант 2 (не нормальный) GSM-1/2-RSP: [0,0] 0D GSM-1/2-RSP: [0,0] 0A GSM-1/2-RSP: [0,0] 2B GSM-1/2-RSP: [0,1] 57 GSM-1/2-RSP: [0,2] 49 GSM-1/2-RSP: [0,3] 4E GSM-1/2-RSP: [0,4] 44 GSM-1/2-RSP: [0,5] 3A GSM-1/2-RSP: [0,6] 20 GSM-1/2-RSP: [0,7] 35 GSM-1/2-RSP: [0,8] 2C GSM-1/2-RSP: [0,9] 31 GSM-1/2-RSP: [0,10] 0D GSM-1/2-RSP: [0,10] 0A [3227.368] GSM-1/2: RECV, Line 0 : +WIND: 5,1 [3227.368] GSM-1/2: RECV, Line 0 : 2B 57 49 4E 44 3A 20 35 2C 31 GSM-1/2-RSP: [1,0] 0D GSM-1/2-RSP: [1,0] 0A GSM-1/2-RSP: [1,0] 2B GSM-1/2-RSP: [1,1] 57 GSM-1/2-RSP: [1,2] 49 GSM-1/2-RSP: [1,3] 4E GSM-1/2-RSP: [1,4] 44 GSM-1/2-RSP: [1,5] 3A GSM-1/2-RSP: [1,6] 20 GSM-1/2-RSP: [1,7] 32 GSM-1/2-RSP: [1,8] 0D GSM-1/2-RSP: [1,8] 0A [3234.627] GSM-1/2: RECV, Line 1 : +WIND: 2 [3234.627] GSM-1/2: RECV, Line 1 : 2B 57 49 4E 44 3A 20 32 GSM-1/2-RSP: [2,0] 0D GSM-1/2-RSP: [2,0] 0A GSM-1/2-RSP: [2,0] 42 GSM-1/2-RSP: [2,1] 55 GSM-1/2-RSP: [2,2] 53 GSM-1/2-RSP: [2,3] 59 GSM-1/2-RSP: [2,4] 0D GSM-1/2-RSP: [2,4] 0A [3238.043] GSM-1/2: RECV, Line 2 : BUSY [3238.043] GSM-1/2: RECV, Line 2 : 42 55 53 59 GSM-1/2-RSP: [3,0] 0D GSM-1/2-RSP: [3,0] 0A GSM-1/2-RSP: [3,0] 2B GSM-1/2-RSP: [3,1] 57 GSM-1/2-RSP: [3,2] 49 GSM-1/2-RSP: [3,3] 4E GSM-1/2-RSP: [3,4] 44 GSM-1/2-RSP: [3,5] 3A GSM-1/2-RSP: [3,6] 20 GSM-1/2-RSP: [3,7] 36 GSM-1/2-RSP: [3,8] 2C GSM-1/2-RSP: [3,9] 31 GSM-1/2-RSP: [3,10] 0D GSM-1/2-RSP: [3,10] 0A [3238.046] GSM-1/2: RECV, Line 3 : +WIND: 6,1 [3238.046] GSM-1/2: RECV, Line 3 : 2B 57 49 4E 44 3A 20 36 2C 31 GSM-1/2-RSP: [0,0] 0D GSM-1/2-RSP: [0,0] 0A GSM-1/2-RSP: [0,0] 4F GSM-1/2-RSP: [0,1] 4B GSM-1/2-RSP: [0,2] 0D GSM-1/2-RSP: [0,2] 0A [3238.142] GSM-1/2: RECV, Line 0 : OK [3238.142] GSM-1/2: RECV, Line 0 : 4F 4B GSM-1/2-RSP: [0,0] 0D GSM-1/2-RSP: [0,0] 0A GSM-1/2-RSP: [0,0] 2B GSM-1/2-RSP: [0,1] 43 GSM-1/2-RSP: [0,2] 53 GSM-1/2-RSP: [0,3] 51 GSM-1/2-RSP: [0,4] 3A GSM-1/2-RSP: [0,5] 20 GSM-1/2-RSP: [0,6] 32 GSM-1/2-RSP: [0,7] 31 GSM-1/2-RSP: [0,8] 2C GSM-1/2-RSP: [0,9] 34 GSM-1/2-RSP: [0,10] 0D GSM-1/2-RSP: [0,10] 0A [3244.773] GSM-1/2: RECV, Line 0 : +CSQ: 21,4 [3244.773] GSM-1/2: RECV, Line 0 : 2B 43 53 51 3A 20 32 31 2C 34 GSM-1/2-RSP: [1,0] 0D GSM-1/2-RSP: [1,0] 0A GSM-1/2-RSP: [1,0] 4F GSM-1/2-RSP: [1,1] 4B GSM-1/2-RSP: [1,2] 0D GSM-1/2-RSP: [1,2] 0A [3244.773] GSM-1/2: RECV, Line 1 : OK [3244.773] GSM-1/2: RECV, Line 1 : 4F 4B

Автор:	Denis [ 21 май 2012, 07:10 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
Дебаги нужны одновременно. И по одному порту, чтоб лишние звонки не мешали.

Автор:	Sergius [ 24 май 2012, 12:08 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
Ок Вариант 1 (нормальный) Received SIP PDU from ( 10.202.4.1:5060 ) INVITE sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060> From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 421 v=0 o=3cxPS 378661765120 211023822849 IN IP4 10.202.4.1 s=3cxPS Audio call c=IN IP4 10.202.4.1 t=0 0 m=audio 7094 RTP/AVP 0 8 3 13 9 18 110 99 101 c=IN IP4 10.202.4.1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 iLBC/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 To: <sip:7102@10.202.4.3:5060> Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 1 <Call 585> : *** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337874686) ** 2 <SIP 585> : Receive INVITE Request 3 <NetCon 585> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1) 4 <NetCon 585> : Found inbound voip peer by dest-pattern id(12201) 5 <NetCon 585> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3) 6 <Call 585> : From Net - calledParty(7102) callingParty(10013) 7 <Call 585> : MatchedPerfect 8 <Call 585> : MatchAllProcess After Sorted <0> id(4589) dest(....) prefer(0) selected(7) <1> id(4590) dest(....) prefer(0) selected(7) <2> id(4591) dest(....) prefer(0) selected(7) <3> id(4588) dest(....) prefer(0) selected(8) 9 <Call 585> : Initiate callee with dial-peer(....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 10 <CEP 010100> : InitiateOutCall : calledNum(7102), callingNum(10013), callerPort(ffffffff) type(GSM) [165500.400] RTA(1/1/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1) [165500.405] VP(1/1/0) open channel [165500.405] VM(1/1/0) Tx GSM CallRequest 1 stage 7102 11 <CEP 010100> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(585) [165500.405] VM(1/1/0) set T38 enable by CCC [165500.405] VM(1/1/0) set T38 mode STD [165500.405] VM(1/1/0) Fax rate 9600 12 <SIP 585> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 13 <SIP 585> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 14 <PhonePlay 585> : Audio Count(1) 15 <PhonePlay 585> : rtpSessionId(1) Second Audio Port(-1) 16 <SIP 585> : SetAlerting 17 <Call 585> : PreConnected from(10100) [165500.405] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [165500.405] VM(1/1/0) VAD disable [165500.405] VP(1/1/0) update VAD 0 [165500.405] VM(1/1/0) SID enable by CCC 18 <SIP 585> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 19 <SIP 585> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 20 <SIP 585> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 To: <sip:7102@10.202.4.3:5060>;tag=fe4f374ea4 Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337874686 1337874686 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337874686 0 m=audio 24170 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [165500.415] RTA(1/1/0) Rx RS_OPEN_REQ callId=585 ssId=1 G711U peer=10.202.4.1 mp=24170/24171 hp=7094/7095 [165500.420] VM(1/1/0) codec same G711U [165500.420] RTA(1/1/0) Rx RS_LISTEN_REQ callId=585 ssId=1 G711U peer=10.202.4.1 mp=24170/24171 hp=7094/7095 [165509.205] RTA(1/1/0) Rx GSM_STTS_IND CALL_CONN [165509.205] RTA(1/1/0) Rx GSM_STTS_CALL_CONN at state=5 [165509.205] VP(1/1/0) attribute Fax enable, Modem disable [165509.205] VP(1/1/0) update Fax enable, Modem disable [165509.205] VM(1/1/0) Tx CONNECT_CNF 21 <Call 585> : Connected from(10100) [165509.205] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [165509.205] VM(1/1/0) VAD disable [165509.205] VM(1/1/0) SID enable by CCC 22 <SIP 585> : SetConnected 23 <SIP 585> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 24 <SIP 585> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 25 <SIP 585> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-e108521059601175-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 To: <sip:7102@10.202.4.3:5060>;tag=fe4f374ea4 Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337874695 1337874695 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337874695 0 m=audio 24170 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [165509.220] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [165509.220] VM(1/1/0) DTMF_RTP_RFC2833 enable [165509.220] RTA(1/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65 Received SIP PDU from ( 10.202.4.1:5060 ) ACK sip:7102@10.202.4.3 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-804aad5b2738457d-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060>;tag=fe4f374ea4 From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 1 ACK User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 0 26 <SIP 585> : ACK received 27 <SIP 585> : Receive ACK Request 28 <SIP 585> : Set Terminated Success for 1 INVITE [165512.545] RTA(1/1/0) Rx GSM_STTS_IND CALL_DISC [165512.545] RTA(1/1/0) Rx GSM_STTS_CALL_DISC at state=6 [165512.545] VM(1/1/0) vopp idle [165512.545] VP(1/1/0) close channel [165512.545] VM(1/1/0) Tx DISCONN_CNF 29 <CEP 010100> : Disconnected(16) at Busy 30 <Call 585> : Terminated from(10100) this(Local:CallClear) before((null)) forced(0) time(1337874698) 31 <SIP 585> : ReleaseWithBYE 32 <SIP 585> : Send BYE Request Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 BYE sip:10013@10.202.4.1 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bKfe4f374ea497 From: <sip:7102@10.202.4.3:5060>;tag=fe4f374ea4 To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 97 BYE Date: Thu, 24 May 2012 15:51:38 GMT User-Agent: AddPac SIP Gateway Contact: <sip:7102@10.202.4.3> Content-Length: 0 Max-Forwards: 70 [165512.555] RTA(1/1/0) Rx RS_CLOSE_REQ callId=585 ssId=1 dir=all [165512.555] RTA(1/1/0) close Media socket [165512.555] RTA(1/1/0) close RTCP socket 33 <NetEP 585> : Call FROM <Sergey> terminated reason(Local:CallClear) 34 <CEP 010100> : DisconnectCall at Idle Received SIP PDU from ( 10.202.4.1:5060 ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.202.4.3:5060;branch=z9hG4bKfe4f374ea497 Contact: <sip:10013@10.202.4.1:5060> To: "Sergey"<sip:10013@10.202.4.1:5060>;tag=21508e32 From: <sip:7102@10.202.4.3:5060>;tag=fe4f374ea4 Call-ID: OGYzOGE4ZGYyMTEzMzQxNzcwMmU2OTFjZDQxOTlhNTU. CSeq: 97 BYE User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 0 35 <SIP 585> : Receive 200 OK 36 <SIP 585> : Transaction (97 BYE) completed Вариант 2 (не нормальный) Received SIP PDU from ( 10.202.4.1:5060 ) INVITE sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060> From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00 Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 419 v=0 o=3cxPS 81117839360 21625831425 IN IP4 10.202.4.1 s=3cxPS Audio call c=IN IP4 10.202.4.1 t=0 0 m=audio 7158 RTP/AVP 0 8 3 13 9 18 110 99 101 c=IN IP4 10.202.4.1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 iLBC/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00 To: <sip:7102@10.202.4.3:5060> Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 2 <Call 591> : **** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337875310) ** 3 <SIP 591> : Receive INVITE Request 4 <NetCon 591> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1) 5 <NetCon 591> : Found inbound voip peer by dest-pattern id(12201) 6 <NetCon 591> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3) 7 <Call 591> : From Net - calledParty(7102) callingParty(10013) 8 <Call 591> : MatchedPerfect 9 <Call 591> : MatchAllProcess After Sorted <0> id(4591) dest(....) prefer(0) selected(8) <1> id(4588) dest(....) prefer(0) selected(9) <2> id(4589) dest(....) prefer(0) selected(9) <3> id(4590) dest(....) prefer(0) selected(9) 10 <Call 591> : Initiate callee with dial-peer(....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 11 <CEP 010300> : InitiateOutCall : calledNum(7102), callingNum(10013), callerPort(ffffffff) type(GSM) [166127.515] RTA(1/3/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1) [166127.515] VP(1/3/0) open channel [166127.515] VM(1/3/0) Tx GSM CallRequest 1 stage 7102 12 <CEP 010300> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(591) [166127.515] VM(1/3/0) set T38 enable by CCC [166127.515] VM(1/3/0) set T38 mode STD [166127.520] VM(1/3/0) Fax rate 9600 13 <SIP 591> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 14 <SIP 591> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 15 <PhonePlay 591> : Audio Count(1) 16 <PhonePlay 591> : rtpSessionId(1) Second Audio Port(-1) 17 <SIP 591> : SetAlerting 18 <Call 591> : PreConnected from(10300) [166127.520] RTA(1/3/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [166127.520] VM(1/3/0) VAD disable [166127.520] VP(1/3/0) update VAD 0 [166127.520] VM(1/3/0) SID enable by CCC 19 <SIP 591> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 20 <SIP 591> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 21 <SIP 591> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00 To: <sip:7102@10.202.4.3:5060>;tag=6e4f3554a4 Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Content-Type: application/sdp Content-Length: 240 v=0 o=7102 1337875310 1337875310 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337875310 0 m=audio 24182 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [166127.530] RTA(1/3/0) Rx RS_OPEN_REQ callId=591 ssId=1 G711U peer=10.202.4.1 mp=24182/24183 hp=7158/7159 [166127.530] VM(1/3/0) codec same G711U [166127.530] RTA(1/3/0) Rx RS_LISTEN_REQ callId=591 ssId=1 G711U peer=10.202.4.1 mp=24182/24183 hp=7158/7159 [166136.115] RTA(1/3/0) Rx GSM_STTS_IND CALL_DISC [166136.115] RTA(1/3/0) Rx GSM_STTS_CALL_DISC at state=5 [166136.115] VM(1/3/0) vopp idle [166136.115] VP(1/3/0) close channel [166136.115] VM(1/3/0) Tx DISCONN_CNF 22 <CEP 010300> : Disconnected(16) at Busy 23 <Call 591> : Terminated from(10300) this(Local:CallClear) before((null)) forced(0) time(1337875319) Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00 To: <sip:7102@10.202.4.3:5060>;tag=6e4f3554a4 Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 [166136.115] RTA(1/3/0) Rx RS_CLOSE_REQ callId=591 ssId=1 dir=all [166136.115] RTA(1/3/0) close Media socket [166136.115] RTA(1/3/0) close RTCP socket 24 <NetEP 591> : Call FROM <Sergey> terminated reason(Local:CallClear) 25 <CEP 010300> : DisconnectCall at Idle Received SIP PDU from ( 10.202.4.1:5060 ) ACK sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-4678e870be074e64-1---d8754z-;rport Max-Forwards: 70 To: <sip:7102@10.202.4.3:5060>;tag=6e4f3554a4 From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=74195a00 Call-ID: MTc4NjQ0NmVmNTlkMTg5MDY1YTg3ZWRjZDZjODJmYjE. CSeq: 1 ACK Content-Length: 0 26 <SIP 591> : Receive ACK Request Received SIP PDU from ( 10.202.4.1:5060 ) INVITE sip:7102@10.202.4.3:5060 SIP/2.0 Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-192e2717ef119f79-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:10013@10.202.4.1:5060> To: <sip:7102@10.202.4.3:5060> From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=ff61ff1b Call-ID: NmM4Y2VhMDcyMmNmZmMwNjA2YjJiNzJmZjg5YmRkNmY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE Content-Type: application/sdp Supported: replaces User-Agent: 3CXPhoneSystem 10.0.22539.0 Content-Length: 419 v=0 o=3cxPS 6593445888 454327009281 IN IP4 10.202.4.1 s=3cxPS Audio call c=IN IP4 10.202.4.1 t=0 0 m=audio 7160 RTP/AVP 0 8 3 13 9 18 110 99 101 c=IN IP4 10.202.4.1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:13 CN/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 iLBC/8000 a=rtpmap:99 SPEEX/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-192e2717ef119f79-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=ff61ff1b To: <sip:7102@10.202.4.3:5060> Call-ID: NmM4Y2VhMDcyMmNmZmMwNjA2YjJiNzJmZjg5YmRkNmY. CSeq: 1 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 27 <Call 592> : *** Call Created status(InitiatedByNet) ver(8.28:2006-02-06-00-00) time(1337875319) ** 28 <SIP 592> : Receive INVITE Request 29 <NetCon 592> : Found inbound voip peer(12201) result(3) peer->fixedPatternSize(3) mostMatchingSize(-1) 30 <NetCon 592> : Found inbound voip peer by dest-pattern id(12201) 31 <NetCon 592> : Found inbound voip peer(12202) result(3) peer->fixedPatternSize(1) mostMatchingSize(3) 32 <Call 592> : From Net - calledParty(7102) callingParty(10013) 33 <Call 592> : MatchedPerfect 34 <Call 592> : MatchAllProcess After Sorted <0> id(4588) dest(....) prefer(0) selected(9) <1> id(4589) dest(....) prefer(0) selected(9) <2> id(4590) dest(....) prefer(0) selected(9) <3> id(4591) dest(....) prefer(0) selected(9) 35 <Call 592> : Initiate callee with dial-peer(....) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 36 <CEP 010000> : InitiateOutCall : calledNum(7102), callingNum(10013), callerPort(ffffffff) type(GSM) [166136.245] RTA(1/0/0) Rx CC_OFFHOOK_REQ [2a 37 31 30 32 ] peerId(-1) [166136.245] VP(1/0/0) open channel [166136.245] VM(1/0/0) Tx GSM CallRequest 1 stage 7102 37 <CEP 010000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(592) [166136.250] VM(1/0/0) set T38 enable by CCC [166136.250] VM(1/0/0) set T38 mode STD [166136.250] VM(1/0/0) Fax rate 9600 38 <SIP 592> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 39 <SIP 592> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 40 <PhonePlay 592> : Audio Count(1) 41 <PhonePlay 592> : rtpSessionId(1) Second Audio Port(-1) 42 <SIP 592> : SetAlerting 43 <Call 592> : PreConnected from(10000) [166136.250] RTA(1/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 VAD_CTRL 0 [166136.250] VM(1/0/0) VAD disable [166136.250] VP(1/0/0) update VAD 0 [166136.250] VM(1/0/0) SID enable by CCC 44 <SIP 592> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE) 45 <SIP 592> : SetLocalAudioFormats : myVoipPeer(12201) is not NULL, voiceCodecClass(10) 46 <SIP 592> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( 10.202.4.1:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.202.4.1:5060;branch=z9hG4bK-d8754z-192e2717ef119f79-1---d8754z-;rport From: "Sergey"<sip:10013@10.202.4.1:5060>;tag=ff61ff1b To: <sip:7102@10.202.4.3:5060>;tag=774fa155a4 Call-ID: NmM4Y2VhMDcyMmNmZmMwNjA2YjJiNzJmZjg5YmRkNmY. CSeq: 1 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:7102@10.202.4.3 Content-Type: application/sdp Content-Length: 240 v=0 o=*7102 1337875319 1337875319 IN IP4 10.202.4.3 s=AddPac Gateway SDP c=IN IP4 10.202.4.3 t=1337875319 0 m=audio 24184 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv [166136.260] RTA(1/0/0) Rx RS_OPEN_REQ callId=592 ssId=1 G711U peer=10.202.4.1 mp=24184/24185 hp=7160/7161 [166136.260] VM(1/0/0) codec same G711U [166136.265] RTA(1/0/0) Rx RS_LISTEN_REQ callId=592 ssId=1 G711U peer=10.202.4.1 mp=24184/24185 hp=7160/7161

Автор:	Sergius [ 25 май 2012, 13:08 ]
Заголовок сообщения:	Re: GS-1004B исходящий звонок
Вот тут viewtopic.php?f=11&t=2501 такую же проблему обсуждали, но вопрос переложили с больной головы на здоровую и благополучно забыли...

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