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ap1005, проблемы с кодеками, SDP http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=205 |
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Автор: | deepwalker [ 26 июн 2008, 11:25 ] |
Заголовок сообщения: | ap1005, проблемы с кодеками, SDP |
Ситуация следующая, есть два шлюза, при звонке с одного на другой голос идет только от вызываемой стороны. вызывающий шлюз отсылает такой sdp: v=0 o=- 1189544700 1189544700 IN IP4 192.168.0.8 s=AddPac Gateway SDP c=IN IP4 192.168.0.8 t=1189544700 0 m=audio 23964 RTP/AVP 18 4 116 2 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:116 G726-16/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Вызываемый отсылает такой: v=0 o=989138657866 1214496470 1214496470 IN IP4 10.24.1.8 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214496470 0 m=audio 23434 RTP/AVP 18 a=rtpmap:18 G729/8000/3 a=ptime:20 Так как по середине стоит Freeswitch, то вызывающий получает в итоге такой: v=0 o=989138657866 4667768711296699587 6633288917781619787 IN IP4 192.168.0.198 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214496173 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 G729/8000/3 Меня терзают подозрения, что FS не нравится G729/8000/3, а шлюз не воспринимает rtpmap:96 G729/8000/3 Нет ли способа заставить Addpac отсылать G729/8000/1? |
Автор: | Geniu$$ [ 26 июн 2008, 14:15 ] |
Заголовок сообщения: | |
Кодек какой стоит у вас в диал пире на адпаке? Лучше покажите конфиги с адапаков целиком. И дебаг во время звонка, команды: deb voip call deb rta ipc deb voip sip conf t deb |
Автор: | deepwalker [ 26 июн 2008, 16:37 ] |
Заголовок сообщения: | |
Первый, с него звоню: Код: ! version 8.30I ! hostname AP1005 ! ! no bridge spanning-tree ! ! no ip-share enable ! interface ether0.0 ip address 192.168.0.8 255.255.255.0 ! interface ether1.0 no ip address ! snmp name AP1005 ! no arp reset ! route 0.0.0.0 0.0.0.0 192.168.0.5 ! ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol t38 redundancy 0 fax rate 9600 h323 call start fast h323 call tunnel enable translate-voip-incoming called-number 0 ! ! ! Voice port configuration. ! ! FXO voice-port 0/0 no caller-id enable ! ! ! FXO voice-port 0/1 no caller-id enable ! ! ! FXO voice-port 0/2 no caller-id enable ! ! ! FXO voice-port 0/3 no caller-id enable ! ! ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 100T port 0/0 ! dial-peer voice 1 pots destination-pattern 100T port 0/1 ! dial-peer voice 2 pots destination-pattern 100T port 0/2 ! dial-peer voice 3 pots destination-pattern 100T port 0/3 ! dial-peer voice 4 pots destination-pattern 1001 port 0/0 ! ! ! ! Voip peer configuration. ! dial-peer voice 1000 voip destination-pattern T session target 192.168.0.198 5061 session protocol sip voice-class codec 1 dtmf-relay info vad ! ! Codec classes configuration. ! voice class codec 1 codec preference 1 g729 codec preference 2 g7231r53 codec preference 3 g7231r63 codec preference 4 g726r16 codec preference 5 g726r32 codec preference 6 g711ulaw codec preference 7 g711alaw ! ! ! ! Translation Rule configuration. ! translation-rule 0 rule 0 T 100 ! ! SIP UA configuration. ! sip-ua srv enable ! ! Tones voice class clear-down-cadence 1 -10 350 330 5 11 ! Второй, на него Код: !
version 8.30I ! hostname AP1005 ! ! no bridge spanning-tree ! ! no ip-share enable ! interface ether0.0 ip address 10.24.1.8 255.255.255.0 ! interface ether1.0 no ip address ! snmp name AP1005 ! no arp reset ! route 0.0.0.0 0.0.0.0 10.24.1.5 ! ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol t38 redundancy 0 fax rate 9600 h323 call start fast h323 call tunnel enable ! ! ! Voice port configuration. ! ! FXO voice-port 0/0 connection plar 805 description city no caller-id enable ! ! ! ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 9T port 0/0 ! ! Voip peer configuration. ! dial-peer voice 1000 voip destination-pattern T session target 10.24.1.1 session protocol sip voice-class codec 1 dtmf-relay rtp-2833 vad ! ! Codec classes configuration. ! voice class codec 1 codec preference 1 g729 codec preference 2 g726r16 codec preference 3 g726r32 codec preference 4 g7231r53 codec preference 5 g7231r63 codec preference 6 g711alaw codec preference 7 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua sip-username 200 sip-server 10.24.1.1 set-local-domain 10.24.1.1 ! ! Tones voice class clear-down-cadence 1 -14 170 170 5 0 ! Отладка будет завтра, из дома сложно позвонить через шлюз. |
Автор: | deepwalker [ 27 июн 2008, 02:27 ] |
Заголовок сообщения: | |
Отладка со второго addpac (на который звоню): Код: Received SIP PDU from ( 192.168.0.198:5061 )
INVITE sip:989138657866@10.24.1.8 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKDF2UgHZe0vUHj Max-Forwards: 69 From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8> Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148378 INVITE Contact: <sip:mod_sofia@192.168.0.198:5061> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, PUBLISH Supported: 100rel, precondition, timer Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 378 Remote-Party-ID: "unknown" <sip:@cluecon.com>;screen=yes;privacy=off v=0 o=- 2359798609784629765 4960854064931348180 IN IP4 192.168.0.198 s=AddPac Gateway SDP c=IN IP4 192.168.0.8 t=1189607155 0 m=audio 23020 RTP/AVP 18 4 116 2 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:116 G726-16/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKDF2UgHZe0vUHj From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8> Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148378 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 1 <Call 8241> : ****************** Call Created status(InitiatedByNet) ******************* 2 <SIP 8241> : Receive INVITE Request 3 <NetCon 8241> : Can not found suitable inbound voip peer - apply default 4 <Call 8241> : From Net - calledParty(989138657866) callingParty() 5 <Call 8241> : MatchedAll 6 <Call 8241> : MatchAllProcess After Sorted <0> id(0) dest(9T) prefer(0) selected(3830) 7 <Call 8241> : Initiate callee with dial-peer(9T) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 8 <CEP 000000> : InitiateOutCall : calledNum(89138657866), callingNum(), callerPort(ffffffff) type(FXO) [8606387.710] RTA(0/0/0) Rx CC_OFFHOOK_REQ [38 39 31 33 38 36 35 37 38 36 36 ] peerId(-1) [8606387.710] VM(0/0/0) FXO OffHook [8606387.710] VM(0/0/0) vopp enable 9 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(8241) [8606387.710] VM(0/0/0) set T38 mode STD [8606387.710] VM(0/0/0) Fax rate 9600 10 <SIP 8241> : SetAlerting 11 <Call 8241> : PreConnected from(0) 12 <SIP 8241> : Add Local Audio MediaFormat : 18 Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKDF2UgHZe0vUHj From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148378 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:989138657866@10.24.1.8 RSeq: 573290 Require: 100rel Content-Type: application/sdp Content-Length: 181 v=0 o=989138657866 1214558628 1214558628 IN IP4 10.24.1.8 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214558628 0 m=audio 23482 RTP/AVP 18 a=rtpmap:18 G729/8000/3 a=ptime:20 [8606387.740] RTA(0/0/0) Rx RS_OPEN_REQ callId=8241 ssId=1 G729A peer=192.168.0.8 mp=23482/23483 hp=23020/23021 [8606387.740] VM(0/0/0) codec same G729A Received SIP PDU from ( 192.168.0.198:5061 ) PRACK sip:989138657866@10.24.1.8 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKerUmjcgjX5H4D Max-Forwards: 70 From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148379 PRACK Contact: <sip:mod_sofia@192.168.0.198:5061> RAck: 573290 101148378 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 13 <SIP 8241> : Receive PRACK Request Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKerUmjcgjX5H4D From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148379 PRACK User-Agent: AddPac SIP Gateway Content-Length: 0 [8606388.040] VM(0/0/0) CDTC tone actv detected [8606388.040] VM(0/0/0) CDTC pBuf[17]: 14 14 14 14 15 14 15 14 15 14 15 14 15 14 15 14 14 [8606388.210] VM(0/0/0) CDTC tone actv ignore [8606388.210] VM(0/0/0) CDTC pBuf[17]: 15 15 15 15 14 15 14 15 14 15 14 15 14 15 14 15 15 [8606388.380] VM(0/0/0) CDTC tone actv detected [8606388.380] VM(0/0/0) CDTC pBuf[17]: 14 14 14 14 15 14 15 14 15 14 15 14 15 14 15 14 14 [8606388.550] VM(0/0/0) CDTC tone actv ignore [8606388.550] VM(0/0/0) CDTC pBuf[17]: 15 15 15 15 14 15 14 15 14 15 14 15 14 15 14 15 15 [8606388.710] VM(0/0/0) play digit '8' [8606388.720] VM(0/0/0) CDTC tone actv detected [8606388.720] VM(0/0/0) CDTC pBuf[17]: 14 14 14 14 15 14 15 14 15 14 15 14 15 14 15 14 14 [8606388.860] VM(0/0/0) play mute [8606388.890] VM(0/0/0) CDTC tone idle detected [8606388.890] VM(0/0/0) CDTC pBuf[17]: 60 60 60 60 60 60 60 60 60 60 60 60 60 60 60 60 60 [8606388.960] VM(0/0/0) play digit '9' [8606389.060] VM(0/0/0) CDTC tone idle ignore [8606389.060] VM(0/0/0) CDTC pBuf[17]: 60 60 60 60 15 14 15 14 60 60 60 60 60 60 60 60 60 [8606389.110] VM(0/0/0) play mute [8606389.210] VM(0/0/0) play digit '1' [8606389.360] VM(0/0/0) play mute [8606389.460] VM(0/0/0) play digit '3' [8606389.610] VM(0/0/0) play mute [8606389.710] VM(0/0/0) play digit '8' [8606389.860] VM(0/0/0) play mute [8606389.960] VM(0/0/0) play digit '6' [8606390.110] VM(0/0/0) play mute [8606390.210] VM(0/0/0) play digit '5' [8606390.360] VM(0/0/0) play mute [8606390.460] VM(0/0/0) play digit '7' [8606390.610] VM(0/0/0) play mute [8606390.710] VM(0/0/0) play digit '8' [8606390.860] VM(0/0/0) play mute [8606390.960] VM(0/0/0) play digit '6' [8606391.110] VM(0/0/0) play mute [8606391.210] VM(0/0/0) play digit '6' [8606391.360] VM(0/0/0) play mute [8606391.460] VM(0/0/0) Fax enable [8606391.460] VM(0/0/0) play mute [8606391.460] VM(0/0/0) Tx CONNECT_CNF 14 <Call 8241> : Connected from(0) 15 <SIP 8241> : SetConnected 16 <SIP 8241> : Add Local Audio MediaFormat : 18 Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKDF2UgHZe0vUHj From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148378 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:989138657866@10.24.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Length: 0 [8606391.480] RTA(0/0/0) Rx RS_LISTEN_REQ callId=8241 ssId=1 G729A peer=192.168.0.8 mp=23482/23483 hp=23020/23021 [8606391.480] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 0 [8606391.480] VM(0/0/0) DTMF disable [8606391.485] VM(0/0/0) play mute Received SIP PDU from ( 192.168.0.198:5061 ) ACK sip:989138657866@10.24.1.8 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKF1mDm70Nte8pS Max-Forwards: 70 From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148378 ACK Contact: <sip:mod_sofia@192.168.0.198:5061> Content-Length: 0 17 <SIP 8241> : ACK received 18 <SIP 8241> : Receive ACK Request 19 <SIP 8241> : Set Terminated Success for 101148378 INVITE [8606397.170] VM(0/0/0) CDTC tone actv detected [8606397.170] VM(0/0/0) CDTC pBuf[17]: 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 [8606397.340] VM(0/0/0) CDTC tone actv ignore [8606397.340] VM(0/0/0) CDTC pBuf[17]: 17 17 17 17 16 17 16 17 16 17 16 17 16 16 16 17 17 [8606397.510] VM(0/0/0) CDTC tone actv detected [8606397.510] VM(0/0/0) CDTC pBuf[17]: 16 16 16 16 17 16 16 16 16 16 16 16 16 16 16 16 16 [8606397.680] VM(0/0/0) CDTC tone actv ignore [8606397.680] VM(0/0/0) CDTC pBuf[17]: 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 [8606397.850] VM(0/0/0) CDTC tone actv detected [8606397.850] VM(0/0/0) CDTC pBuf[17]: 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 [8606398.020] VM(0/0/0) CDTC tone actv ignore [8606398.020] VM(0/0/0) CDTC pBuf[17]: 16 16 16 16 16 16 16 16 16 16 16 16 16 16 16 18 18 Received SIP PDU from ( 192.168.0.198:5061 ) BYE sip:989138657866@10.24.1.8 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKgae6N2HSQQy9m Max-Forwards: 70 From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148380 BYE Contact: <sip:mod_sofia@192.168.0.198:5061> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, PUBLISH Supported: 100rel, precondition, timer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 20 <SIP 8241> : Receive BYE Request Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKgae6N2HSQQy9m From: "unknown" <sip:192.168.0.198>;tag=cmeHKeZ0S2HHc To: <sip:989138657866@10.24.1.8>;tag=a44854a4a4 Call-ID: eb1f79b0-be92-122b-7a87-001676c94d0e CSeq: 101148380 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 21 <SIP 8241> : ReleaseWithNothing [8606405.685] RTA(0/0/0) Rx RS_CLOSE_REQ callId=8241 ssId=1 dir=reve [8606405.685] RTA(0/0/0) Rx RS_CLOSE_REQ callId=8241 ssId=1 dir=forw [8606405.685] RTA(0/0/0) close Media socket [8606405.685] RTA(0/0/0) close RTCP socket 22 <Call 8241> : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0) 23 <CEP 000000> : DisconnectCall at Busy 24 <CEP 000000> : StopSignal [8606405.685] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_STOP [8606405.685] VM(0/0/0) play mute 25 <CEP 000000> : Disconnect (0) [8606405.685] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0) [8606405.685] VM(0/0/0) vopp idle [8606405.685] VM(0/0/0) FXO OnHook [8606405.690] VM(0/0/0) Tx DISCONN_CNF 26 <NetEP 8241> : Call TO <unknown> terminated reason(Remote:CallClear) 27 <CEP 000000> : Disconnected(16) at Disconnecting |
Автор: | deepwalker [ 27 июн 2008, 03:07 ] |
Заголовок сообщения: | |
Вызов идет через этот шлюз: Код: [4559.245] VM(0/0/0) Rx FXO Ring Actv
[4559.245] VM(0/0/0) Tx RING_IND 1 <CEP 000000> : Call Received [4559.755] VM(0/0/0) Rx FXO Ring Idle [4559.755] VM(0/0/0) FXO OffHook [4559.755] VM(0/0/0) vopp enable [4559.755] VM(0/0/0) play Dial tone [4559.755] VM(0/0/0) Tx OFFHOOK_IND 2 <CEP 000000> : Call Initiated : calledNumber() crv(0) total(0) 3 <Call 30> : ****************** Call Created status(InitiatedByFXO) ******************* 4 <CEP 000000> : Calling number() 5 <CEP 000000> : Call id(89fde746-7a17-a321-8028-0002a4049b78) callNum(30) [4559.765] VM(0/0/0) Rx FXO Ring Actv ignore [4559.855] VM(0/0/0) Rx FXO Ring Actv ignore [4559.875] VM(0/0/0) Rx FXO Ring Actv [4559.905] VM(0/0/0) Rx FXO Ring Idle [4562.175] VM(0/0/0) Tx DIGIT_IND '3' [4562.180] VM(0/0/0) play mute 6 <Call 30> : Digit(3) at InitiatedByFXO 7 <Call 30> : MatchedAll [4562.375] VM(0/0/0) Tx DIGIT_IND '9' 8 <Call 30> : Digit(9) at CalleeDeterminedWaitDigit 9 <Call 30> : MatchedAll [4562.575] VM(0/0/0) Tx DIGIT_IND '8' 10 <Call 30> : Digit(8) at CalleeDeterminedWaitDigit 11 <Call 30> : MatchedAll [4562.775] VM(0/0/0) Tx DIGIT_IND '9' 12 <Call 30> : Digit(9) at CalleeDeterminedWaitDigit 13 <Call 30> : MatchedAll [4562.975] VM(0/0/0) Tx DIGIT_IND '1' 14 <Call 30> : Digit(1) at CalleeDeterminedWaitDigit 15 <Call 30> : MatchedAll [4563.175] VM(0/0/0) Tx DIGIT_IND '3' 16 <Call 30> : Digit(3) at CalleeDeterminedWaitDigit 17 <Call 30> : MatchedAll [4563.375] VM(0/0/0) Tx DIGIT_IND '8' 18 <Call 30> : Digit(8) at CalleeDeterminedWaitDigit 19 <Call 30> : MatchedAll [4563.575] VM(0/0/0) Tx DIGIT_IND '6' 20 <Call 30> : Digit(6) at CalleeDeterminedWaitDigit 21 <Call 30> : MatchedAll [4563.775] VM(0/0/0) Tx DIGIT_IND '5' 22 <Call 30> : Digit(5) at CalleeDeterminedWaitDigit 23 <Call 30> : MatchedAll [4563.975] VM(0/0/0) Tx DIGIT_IND '7' 24 <Call 30> : Digit(7) at CalleeDeterminedWaitDigit 25 <Call 30> : MatchedAll [4564.175] VM(0/0/0) Tx DIGIT_IND '8' 26 <Call 30> : Digit(8) at CalleeDeterminedWaitDigit 27 <Call 30> : MatchedAll [4564.375] VM(0/0/0) Tx DIGIT_IND '6' 28 <Call 30> : Digit(6) at CalleeDeterminedWaitDigit 29 <Call 30> : MatchedAll [4564.575] VM(0/0/0) Tx DIGIT_IND '6' 30 <Call 30> : Digit(6) at CalleeDeterminedWaitDigit 31 <Call 30> : MatchedAll 32 <Time 30> : Inter digit timer timeout. 33 <Call 30> : Digit(#) at CalleeDeterminedWaitDigit 34 <Call 30> : MatchAllProcess After Sorted <0> id(1000) dest(T) prefer(0) selected(5) 35 <Call 30> : Initiate callee with dial-peer(T) status(CalleeDeterminedAll) id(89fde746-7a17-a321-8028-0002a4049b78) 36 <NetEP 30> : InitiateOutCall: calledNum(3989138657866) callingNum() target(192.168.0.198:5061) 37 <NetEP 30> : DoCall: calledAddr(sip:3989138657866@192.168.0.198:5061) callingAddr() [4567.580] VM(0/0/0) set T38 mode STD [4567.580] VM(0/0/0) Fax rate 9600 38 <SIP 0> : No authentication information available 39 <SIP 30> : Send INVITE Request Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 INVITE sip:3989138657866@192.168.0.198:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK91462929a420 From: <sip:192.168.0.8>;tag=91462929a4 To: <sip:3989138657866@192.168.0.198> Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 20 INVITE Supported: timer, replaces Min-SE: 1800 Date: Wed, 12 Sep 2007 14:54:09 GMT User-Agent: AddPac SIP Gateway Contact: <sip:192.168.0.8> Accept: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 358 Max-Forwards: 70 v=0 o=- 1189608849 1189608849 IN IP4 192.168.0.8 s=AddPac Gateway SDP c=IN IP4 192.168.0.8 t=1189608849 0 m=audio 23060 RTP/AVP 18 4 116 2 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:116 G726-16/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 [4567.630] RTA(0/0/0) Rx RS_LISTEN_REQ callId=30 ssId=1 G711U peer=0.0.0.0 mp=23060/23061 hp=0/0 [4567.635] VM(0/0/0) vopp idle [4567.635] VM(0/0/0) start codec replace timer to G711U Received SIP PDU from ( 192.168.0.198:5061 ) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK91462929a420 From: <sip:192.168.0.8>;tag=91462929a4 To: <sip:3989138657866@192.168.0.198> Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 20 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Content-Length: 0 [4567.635] VM(0/0/0) discard voice under codec replace [4567.645] VM(0/0/0) discard voice under codec replace 40 <SIP 30> : Receive 100 Trying 41 <SIP 30> : Transaction (20 INVITE) proceeding [4567.695] VM(0/0/0) vopp enable [4567.695] VM(0/0/0) codec replaced to G711U Received SIP PDU from ( 192.168.0.198:5061 ) SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK91462929a420 From: <sip:192.168.0.8>;tag=91462929a4 To: <sip:3989138657866@192.168.0.198>;tag=Fg3mH1cvmtXZH Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 20 INVITE Contact: <sip:mod_sofia@192.168.0.198:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, PUBLISH Supported: 100rel, precondition, timer Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Content-Type: application/sdp Content-Disposition: session Content-Length: 187 v=0 o=989138657866 1090535811911224245 1960318169895965664 IN IP4 192.168.0.198 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214560324 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 G729/8000/3 42 <SIP 30> : Receive 183 Session Progress 43 <SIP 30> : Transaction (20 INVITE) proceeding 44 <SIP 30> : Received Session Progress response 45 <NetCon 30> : Alert received (inband tone explicitly). 46 <Call 30> : Alert from(fffffffe) pseudo(0) inband(1) status(CalleeInitiated) [4571.595] VM(0/0/0) vopp idle [4571.595] VM(0/0/0) start codec replace timer to G729A [4571.595] VM(0/0/0) Rx RTP replace codec to G729A [4571.605] VM(0/0/0) discard voice under codec replace Received SIP PDU from ( 192.168.0.198:5061 ) SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK91462929a420 From: <sip:192.168.0.8>;tag=91462929a4 To: <sip:3989138657866@192.168.0.198>;tag=Fg3mH1cvmtXZH Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 20 INVITE Contact: <sip:mod_sofia@192.168.0.198:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, PUBLISH Supported: 100rel, precondition, timer Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 187 v=0 o=989138657866 1090535811911224245 1960318169895965664 IN IP4 192.168.0.198 s=AddPac Gateway SDP c=IN IP4 10.24.1.8 t=1214560324 0 m=audio 0 RTP/AVP 96 a=rtpmap:96 G729/8000/3 [4571.655] VM(0/0/0) vopp enable [4571.655] VM(0/0/0) codec replaced to G729A 47 <SIP 30> : Receive 200 OK 48 <Call 30> : Connected from(fffffffe) [4571.690] RTA(0/0/0) Rx CC_CONNECT_RSP peerId(0/0/0) [4571.690] VM(0/0/0) Fax enable [4571.690] VM(0/0/0) play mute 49 <NetEP 30> : Call with sip:3989138657866@192.168.0.198 established 50 <SIP 30> : Received INVITE OK response 51 <SIP 30> : Send ACK Request Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 ACK sip:mod_sofia@192.168.0.198:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK91462929a420 From: <sip:192.168.0.8>;tag=91462929a4 To: <sip:3989138657866@192.168.0.198>;tag=Fg3mH1cvmtXZH Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 20 ACK Content-Length: 0 Max-Forwards: 70 52 <SIP 30> : Check Event Relation 53 <SIP 30> : Set Terminated Success for 20 INVITE [4578.180] VM(0/0/0) CDTC tone actv detected [4578.180] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 12 14 12 11 13 14 15 12 13 11 12 10 10 09 09 09 11 09 10 08 08 13 12 11 10 07 07 05 05 07 07 [4578.510] VM(0/0/0) CDTC tone actv ignore [4578.510] VM(0/0/0) CDTC pBuf[33]: 08 08 08 08 09 10 10 13 12 10 15 21 23 28 28 28 26 29 29 31 30 32 32 32 34 33 35 35 35 34 35 34 34 [4582.120] VM(0/0/0) CDTC pBuf[35]: 02 02 02 02 02 03 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 [4582.130] VM(0/0/0) CDTC pBuf[35]: 02 02 02 02 03 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 [4582.140] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 [4582.150] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 [4582.160] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 [4582.170] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 [4582.180] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 [4582.190] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 [4582.200] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 02 [4582.210] VM(0/0/0) CDTC pBuf[35]: 04 04 04 04 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 02 02 [4582.220] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 02 02 02 [4582.230] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 02 02 02 02 [4582.240] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 02 02 02 02 02 [4582.250] VM(0/0/0) CDTC pBuf[35]: 03 03 03 03 03 02 02 03 02 02 02 01 03 03 04 03 03 02 02 03 02 02 02 02 02 02 02 02 02 02 02 02 02 02 02 Received SIP PDU from ( 192.168.0.198:5061 ) BYE sip:192.168.0.8 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKHNSH90ryUt68N Max-Forwards: 70 From: <sip:3989138657866@192.168.0.198>;tag=Fg3mH1cvmtXZH To: <sip:192.168.0.8>;tag=91462929a4 Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 101149234 BYE Contact: <sip:mod_sofia@192.168.0.198:5061;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8833 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO, PUBLISH Supported: 100rel, precondition, timer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 54 <SIP 30> : Receive BYE Request Sending SIP PDU to ( 192.168.0.198:5061 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.198:5061;rport;branch=z9hG4bKHNSH90ryUt68N From: <sip:3989138657866@192.168.0.198>;tag=Fg3mH1cvmtXZH To: <sip:192.168.0.8>;tag=91462929a4 Call-ID: 91fde746-a666-297d-8029-0002a4049b78@192.168.0.8 CSeq: 101149234 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 55 <SIP 30> : ReleaseWithNothing [4584.305] RTA(0/0/0) Rx RS_CLOSE_REQ callId=30 ssId=1 dir=reve [4584.305] RTA(0/0/0) close Media socket [4584.305] RTA(0/0/0) close RTCP socket 56 <Call 30> : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0) 57 <CEP 000000> : DisconnectCall at Busy 58 <CEP 000000> : StopSignal [4584.305] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_STOP [4584.305] VM(0/0/0) play mute 59 <CEP 000000> : Disconnect (0) [4584.305] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0) [4584.310] VM(0/0/0) vopp idle [4584.310] VM(0/0/0) FXO OnHook [4584.310] VM(0/0/0) Tx DISCONN_CNF 60 <NetEP 30> : Call TO <sip:3989138657866@192.168.0.198> terminated reason(Remote:CallClear) 61 <CEP 000000> : Disconnected(16) at Disconnecting |
Автор: | Geniu$$ [ 27 июн 2008, 07:42 ] |
Заголовок сообщения: | |
Добавьте на шлюзы в voip пиры: codec-variant g729 annex-a На втором адпаке dial-p v 666 v codec g729 codec-variant g729 annex-a ses tar 192.168.0.198 |
Автор: | deepwalker [ 27 июн 2008, 10:57 ] |
Заголовок сообщения: | |
Сдается мне косячит freeswitch - зачем надо SDP переписывать? Ваш совет не помог 3 убрать из SDP к сожалению : ( |
Автор: | Geniu$$ [ 27 июн 2008, 12:13 ] |
Заголовок сообщения: | |
Попробуйте на втором адпаке ещё conf t sip response alert with-sdp |
Автор: | deepwalker [ 29 июн 2008, 11:52 ] |
Заголовок сообщения: | |
Попробую в понедельник. Правда мне ребята из команды FS уже дали патчик для обхода проблемы, но только в виде грязного хака - просто стирает /3 на конце. |
Автор: | deepwalker [ 30 июн 2008, 07:22 ] |
Заголовок сообщения: | |
Ничего не помогло : ( А не знаете, как выйти на поддержку addpac подобную русскому форуму dlink? Мне сильно кажется, что это баг. |
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