СвязьПроект :: Просмотр темы - AddPac FXO и передача DTMF в PSTN

! Voice service voip configuration.
!
voice service voip
fax protocol bypass
fax rate 14400
h323 call start fast
h323 call tunnel enable
voice-confirmed-connect 25
timeout tidt 10
timeout tterm 14400
busyout monitor sip-server
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
connection plar 4004
ring detect-timeout 50
ring detect-timer 500
no comfort-noise
high-dtmf-gain 0
caller-id enable
caller-id type etsi
caller-id name disable
forced-clear-down -55 30
!
!
! Pots peer configuration.
!
dial-peer voice 100 pots
destination-pattern T
port 0/0
forward-digits from 0
no register e164
!
!
! Voip peer configuration.
!
dial-peer voice 0 voip
destination-pattern 4004F
session target asterisk
session protocol sip
voice-class codec 1
no vad
dtmf-relay rtp-2833
huntstop
!
!
dial-peer hunt 1
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.10.10.10
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g711alaw
!
!
!
! SIP UA configuration.
!
sip-ua
sip-server asterisk
timeout tregtry 60
register e164
!
!
! MGCP configuration.
!
mgcp
dtmf-relay rtp-2833
codec g711alaw
no vad
!
!
! Tones
voice class clear-down-tone 0 450 0 350 350
!
voice class clear-down-cadence 1 -14 350 350 5 11
!

Автор:	VX [ 02 дек 2008, 17:09 ]
Заголовок сообщения:	AddPac FXO и передача DTMF в PSTN
Есть AddPac AP1100F 8FXO (Firmware 8.30R) + Asterisk + Linksys SPA 941. Все работает по SIP через Asterisk (dtmfmode=rfc2833). Существует проблема с передачей dtmf в PSTN при звонке с IP-телефонов (только в таком направлении), а именно dtmf до шлюза доходят, но дальше в аналоговую линию он их не генерирует . При этом передача dtmf как внутренним так и внешним абонентам при звонках из PSTN работает отлично. Настройки шлюза: Код: ! Voice service voip configuration. ! voice service voip fax protocol bypass fax rate 14400 h323 call start fast h323 call tunnel enable voice-confirmed-connect 25 timeout tidt 10 timeout tterm 14400 busyout monitor sip-server busyout monitor voip-interface ! ! ! Voice port configuration. ! ! FXO voice-port 0/0 connection plar 4004 ring detect-timeout 50 ring detect-timer 500 no comfort-noise high-dtmf-gain 0 caller-id enable caller-id type etsi caller-id name disable forced-clear-down -55 30 ! ! ! Pots peer configuration. ! dial-peer voice 100 pots destination-pattern T port 0/0 forward-digits from 0 no register e164 ! ! ! Voip peer configuration. ! dial-peer voice 0 voip destination-pattern 4004F session target asterisk session protocol sip voice-class codec 1 no vad dtmf-relay rtp-2833 huntstop ! ! dial-peer hunt 1 ! ! gatekeeper ! ! ! Gateway configuration. ! gateway h323-id voip.10.10.10.10 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 1 codec preference 1 g711alaw ! ! ! ! SIP UA configuration. ! sip-ua sip-server asterisk timeout tregtry 60 register e164 ! ! ! MGCP configuration. ! mgcp dtmf-relay rtp-2833 codec g711alaw no vad ! ! ! Tones voice class clear-down-tone 0 450 0 350 350 ! voice class clear-down-cadence 1 -14 350 350 5 11 ! Спасибо за любую помощь.

Автор:	Denis [ 03 дек 2008, 15:02 ]
Заголовок сообщения:	Re: AddPac FXO и передача DTMF в PSTN
Добрый день. Пришлите deb voip call, deb voip sip, deb rta ipc.

Автор:	Николаев Игорь [ 25 май 2009, 15:49 ]
Заголовок сообщения:	Не передаётся dtmf в pstn
Когда ставлю dtmf-relay rtp-2833 - не передаются цифры 1-3. Когда ставлю h245-alphanumeric - передаются все, но очень плохо. Пробовал играться с настройками fxo: output-gain, input-gane, low-dtmf-gain, high-dtmf-gain. Ничего не получается. AP200# sh run ! version 8.23I ! hostname AP200 ! ! no bridge spanning-tree ! dhcp-list 1 type server dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128 ! ! ip-share enable ip-share interface net-side ether0.0 ip-share interface local-side ether1.0 ! interface ether0.0 ip address 192.168.17.111 255.255.255.0 ! interface ether1.0 no ip address ! snmp name AP200E no snmp trap-authentication no snmp enable-trap link-updown no snmp enable-trap cold-warm-start snmp enable-trap dn-register 300 ! no arp reset ! route 0.0.0.0 0.0.0.0 192.168.17.200 ! dnshost nameserver 192.168.17.200 ! ! ! ! ! ! VoIP configuration. ! ! ! Voice service voip configuration. ! voice service voip fax protocol bypass fax rate 14400 h323 call start fast h323 call tunnel enable voice-confirmed-connect 25 ! ! ! Voice port configuration. ! ! FXO voice-port 0/0 no caller-id enable ! ! ! FXS voice-port 0/1 no caller-id enable ! ! ! ! ! Pots peer configuration. ! dial-peer voice 0 pots destination-pattern 1722105 port 0/0 user-password 53244183 ! dial-peer voice 1 pots destination-pattern 1722107 port 0/1 user-password 53244183 ! ! ! ! Voip peer configuration. ! dial-peer voice 100 voip destination-pattern T session target 213.170.81.135 session protocol sip voice-class codec 1 dtmf-relay dual-mode vad ! ! ! ! ! ! ! Gateway configuration. ! gateway h323-id voip.213.170.84.115 no ignore-msg-from-other-gk ! ! ! Codec classes configuration. ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw ! ! ! ! SIP UA configuration. ! sip-ua sip-server 213.170.81.135 register e164 ! ! ! MGCP configuration. ! mgcp codec g711ulaw vad ! ! ! Tones voice class clear-down-cadence 1 -16 350 340 3 11 ! voice class dial-tone 425 0 0 0 0 0 -12 ! voice class ring-back-tone 425 0 1000 4000 0 0 -12 ! voip-interface ether0.0 ! AP200# sh dial- voice 100 VoIP peer 100 dest-pattern = T session-target = 213.170.81.135 session-protocol = SIP answer-address = codec = default voice codec class = 1 dtmfRelay = dual-mode vad = yes sid = yes redundant RTP = no description = preference = 0 huntstop = no translate-outgoing called-number = -1 translate-outgoing calling-number = -1 translate-outgoing digits in call = -1 call diversion = -1 outbound call barred group = -1 fax mode = system fax rate (bps) = system fax T38 redundancy = system max call forward hop = 4 administrative status = up modem passthrough codec = none modem passthrough mode = none AP200# sh voice port 0/0 Voice port slot(0)/port(0) line type = FXO status = Idle input gain = 0 db output gain = 0 db polarity inverse = disabled number of rings to answer = 1 ring detect timeout = 3000 msec ring detect timer = 300 msec FXO hookflash out = 300 msec clear down delay = 0 sec clear down tone detect = enabled tie connection = none description = translate incoming called-number = -1 translate incoming calling-number = -1 comfort noise generation = enabled dial tone generation = enabled echo cancellation = enabled announcement = enabled low dtmf gain = -8 high dtmf gain = -5 caller ID = disabled caller ID type = bellcore caller ID NAME = enabled DID Type = normal busyout action = none backup busyout action = none current callnumber = -1 holded callnumber = -1

Автор:	Geniu$$ [ 26 май 2009, 07:35 ]
Заголовок сообщения:
От куда идёт передача из IP в "город" или из "города" в IP?

Автор:	Николаев Игорь [ 26 май 2009, 09:10 ]
Заголовок сообщения:	Не передаётся dtmf в pstn (ТСОП)
dtmf не передаётся из IP в город в обратном направлении все супер если нужно могу ещё дебаги выложить? у меня такое впечатление, что я где-то какой-то "маленький флажочек" не установил. Причем чем больше паузы при наборе,, тем вероятнее передача dtmf в телефонную сеть общего пользования

СвязьПроект http://old.xdsl.ru/svpro/

AddPac FXO и передача DTMF в PSTN http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=287	Страница 1 из 1

Автор:	Geniu$$ [ 26 май 2009, 12:25 ]
Заголовок сообщения:
Да. покажите дебаг deb voip call deb rta ipc deb voip sip conf t deb Включите dtmf-relay rtp-2833

Автор:	Николаев Игорь [ 26 май 2009, 13:06 ]
Заголовок сообщения:	включил все 3 дебага
включил все 3 дебага, но даже когда звонка нет, там чё-то сыпет, к сожалению не понимаю что это. Если нужно, могу включить какой-то 1 или 2 дебага и прислать. Ну, и с аппарата в линию dtmf проходит и все такое... Вот такие дебаги: (позвонил на FXO, набрал 1234567890 и положил трубку со стороны IP) И ещё, звонок осуществлялся через SIP-сервер 213.170.81.135 с внутреннего на внутренний номер (если это как-то проясняет ситуацию) conf Enter configuration commands, one per line. End with CNTL/Z AP200(config)# deb AP200(config)# ex AP200# 649 <SIP 36081> : Set Terminated Success for 36081 REGISTER 650 <SIP 36082> : Set Terminated Success for 36082 REGISTER 651 <SIP 36083> : Set Terminated Success for 36083 REGISTER 652 <SIP 36084> : Set Terminated Success for 36084 REGISTER Received SIP PDU from ( 213.170.81.135:5060 ) NOTIFY sip:1722105@192.168.17.111 SIP/2.0 Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK0381e820;rport From: "Anonymous" <sip:Anonymous@213.170.81.135>;tag=as77308106 To: <sip:1722105@192.168.17.111> Contact: <sip:Anonymous@213.170.81.135> Call-ID: 53cd50c060a961ca4949a5064de28352@213.170.81.135 CSeq: 102 NOTIFY User-Agent: VoipNow PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@213.170.81.135 Voice-Message: 0/0 (0/0) Sending SIP PDU to ( 213.170.81.135:5060 ) from 5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK0381e820;rport From: "Anonymous" <sip:Anonymous@213.170.81.135>;tag=as77308106 To: <sip:1722105@192.168.17.111> Call-ID: 53cd50c060a961ca4949a5064de28352@213.170.81.135 CSeq: 102 NOTIFY User-Agent: AddPac SIP Gateway Content-Length: 0 Received SIP PDU from ( 213.170.81.135:5060 ) NOTIFY sip:1722107@192.168.17.111 SIP/2.0 Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK609296db;rport From: "Anonymous" <sip:Anonymous@213.170.81.135>;tag=as712a56c4 To: <sip:1722107@192.168.17.111> Contact: <sip:Anonymous@213.170.81.135> Call-ID: 5308c9af77e95bef6b1d310066dde3bd@213.170.81.135 CSeq: 102 NOTIFY User-Agent: VoipNow PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 94 Messages-Waiting: no Message-Account: sip:asterisk@213.170.81.135 Voice-Message: 0/0 (0/0) Sending SIP PDU to ( 213.170.81.135:5060 ) from 5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK609296db;rport From: "Anonymous" <sip:Anonymous@213.170.81.135>;tag=as712a56c4 To: <sip:1722107@192.168.17.111> Call-ID: 5308c9af77e95bef6b1d310066dde3bd@213.170.81.135 CSeq: 102 NOTIFY User-Agent: AddPac SIP Gateway Content-Length: 0 Received SIP PDU from ( 213.170.81.135:5060 ) INVITE sip:1722105@192.168.17.111 SIP/2.0 Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK1868f816;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111> Contact: <sip:101@213.170.81.135> Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 102 INVITE User-Agent: VoipNow PBX Max-Forwards: 70 Date: Tue, 26 May 2009 12:59:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 9578 9578 IN IP4 213.170.81.135 s=session c=IN IP4 213.170.81.135 t=0 0 m=audio 49192 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Sending SIP PDU to ( 213.170.81.135:5060 ) from 5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK1868f816;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111> Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 102 INVITE User-Agent: AddPac SIP Gateway Content-Length: 0 653 <Call 107> : *************** Call Created status(InitiatedByNet) ***************** 654 <SIP 107> : Receive INVITE Request 655 <NetCon 107> : Found inbound voip peer by dest-pattern id(100) 656 <Call 107> : From Net - calledParty(1722105) callingParty(101) 657 <Call 107> : MatchedPerfect 658 <Call 107> : MatchAllProcess After Sorted <0> id(0) dest(1722105) prefer(0) selected(44) 659 <Call 107> : Initiate callee with dial-peer(1722105) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000) 660 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(101), callerPort(ffffffff) type(FXO) [450479.865] RTA(0/0/0) Rx CC_OFFHOOK_REQ peerId(-1) [450479.865] VM(0/0/0) FXO OffHook [450479.865] VM(0/0/0) vopp enable [450479.865] VM(0/0/0) Tx CONNECT_CNF 661 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(107) [450479.865] VM(0/0/0) Fax rate 14400 [450479.870] VM(0/0/0) Set ModemBypass Codec G711U 662 <SIP 107> : SetAlerting 663 <Call 107> : PreConnected from(0) 664 <SIP 107> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( 213.170.81.135:5060 ) from 5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK1868f816;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111>;tag=ac600f8da4 Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 102 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:1722105@192.168.17.111 Content-Type: application/sdp Content-Length: 216 v=0 o=1722105 0 0 IN IP4 192.168.17.111 s=AddPac Gateway SDP c=IN IP4 192.168.17.111 t=0 0 m=audio 23196 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 [450479.930] RTA(0/0/0) Rx RS_OPEN_REQ callId=107 ssId=1 G711U peer=213.170.81.135 mp=23196/23197 hp=49192/49193 [450479.930] VM(0/0/0) codec same G711U 665 <Call 107> : Connected from(0) 666 <SIP 107> : SetConnected 667 <SIP 107> : Add Local Audio MediaFormat : 0 Sending SIP PDU to ( 213.170.81.135:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK1868f816;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111>;tag=ac600f8da4 Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 102 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:1722105@192.168.17.111 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 216 v=0 o=1722105 0 0 IN IP4 192.168.17.111 s=AddPac Gateway SDP c=IN IP4 192.168.17.111 t=0 0 m=audio 23196 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 a=ptime:20 [450480.005] RTA(0/0/0) Rx RS_LISTEN_REQ callId=107 ssId=1 G711U peer=213.170.81.135 mp=23196/23197 hp=49192/49193 [450480.005] VM(0/0/0) V152_modem TxPT=0x64, RxPT=0x64 [450480.005] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_CTRL 1 [450480.005] VM(0/0/0) DTMF_RTP_RFC2833 enable [450480.005] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65 Received SIP PDU from ( 213.170.81.135:5060 ) ACK sip:1722105@192.168.17.111 SIP/2.0 Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK601857b8;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111>;tag=ac600f8da4 Contact: <sip:101@213.170.81.135> Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 102 ACK User-Agent: VoipNow PBX Max-Forwards: 70 Content-Length: 0 668 <SIP 107> : ACK received 669 <SIP 107> : Receive ACK Request 670 <SIP 107> : Set Terminated Success for 102 INVITE [450480.335] VM(0/0/0) CDTC tone actv detected [450480.335] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 [450480.675] VM(0/0/0) CDTC tone actv ignore [450480.675] VM(0/0/0) CDTC pBuf[34]: 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 [450481.025] VM(0/0/0) CDTC tone actv detected [450481.025] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 [450485.775] VM(0/0/0) play digit single '1' [450485.785] VM(0/0/0) play digit single '1' [450485.795] VM(0/0/0) play digit single '1' [450485.805] RTA(0/0/0) RTP play DTMF stop [450485.805] VM(0/0/0) play mute [450485.835] VM(0/0/0) CDTC tone actv ignore [450485.835] VM(0/0/0) CDTC pBuf[34]: 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 60 60 [450486.195] VM(0/0/0) CDTC tone actv detected [450486.195] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 [450486.275] VM(0/0/0) play digit single '2' [450486.285] VM(0/0/0) play digit single '2' [450486.295] VM(0/0/0) play digit single '2' [450486.305] RTA(0/0/0) RTP play DTMF stop [450486.305] VM(0/0/0) play mute [450486.535] VM(0/0/0) CDTC tone actv ignore [450486.535] VM(0/0/0) CDTC pBuf[34]: 13 13 13 13 13 13 13 13 13 60 60 60 60 60 60 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 [450486.815] VM(0/0/0) play digit single '3' [450486.825] VM(0/0/0) play digit single '3' [450486.835] VM(0/0/0) play digit single '3' [450486.845] RTA(0/0/0) RTP play DTMF stop [450486.845] VM(0/0/0) play mute [450487.235] VM(0/0/0) CDTC tone actv detected [450487.235] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 13 [450487.275] VM(0/0/0) play digit single '4' [450487.285] VM(0/0/0) play digit single '4' [450487.295] VM(0/0/0) play digit single '4' [450487.305] RTA(0/0/0) RTP play DTMF stop [450487.305] VM(0/0/0) play mute [450487.575] VM(0/0/0) CDTC tone actv ignore [450487.575] VM(0/0/0) CDTC pBuf[34]: 13 13 13 13 13 60 60 60 60 60 60 13 13 24 60 60 60 60 52 56 60 60 60 59 60 60 60 60 60 60 54 54 60 60 [450487.815] VM(0/0/0) play digit single '5' [450487.825] VM(0/0/0) play digit single '5' [450487.835] VM(0/0/0) play digit single '5' [450487.845] RTA(0/0/0) RTP play DTMF stop [450487.845] VM(0/0/0) play mute [450488.315] VM(0/0/0) play digit single '6' [450488.325] VM(0/0/0) play digit single '6' [450488.335] VM(0/0/0) play digit single '6' [450488.345] RTA(0/0/0) RTP play DTMF stop [450488.345] VM(0/0/0) play mute [450488.815] VM(0/0/0) play digit single '7' [450488.825] VM(0/0/0) play digit single '7' [450488.835] VM(0/0/0) play digit single '7' [450488.845] RTA(0/0/0) RTP play DTMF stop [450488.845] VM(0/0/0) play mute [450489.335] VM(0/0/0) play digit single '8' [450489.345] VM(0/0/0) play digit single '8' [450489.355] VM(0/0/0) play digit single '8' [450489.365] RTA(0/0/0) RTP play DTMF stop [450489.365] VM(0/0/0) play mute [450489.885] VM(0/0/0) play digit single '9' [450489.895] VM(0/0/0) play digit single '9' [450489.905] VM(0/0/0) play digit single '9' [450489.915] RTA(0/0/0) RTP play DTMF stop [450489.915] VM(0/0/0) play mute [450490.385] VM(0/0/0) play digit single '0' [450490.395] VM(0/0/0) play digit single '0' [450490.405] VM(0/0/0) play digit single '0' [450490.415] RTA(0/0/0) RTP play DTMF stop [450490.415] VM(0/0/0) play mute [450491.745] VM(0/0/0) CDTC pBuf[35]: 15 15 15 15 15 15 16 15 14 15 16 15 14 14 14 14 14 13 15 15 14 14 14 14 13 13 13 13 11 11 11 11 11 11 11 [450491.755] VM(0/0/0) CDTC pBuf[35]: 15 15 15 15 15 16 15 14 15 16 15 14 14 14 14 14 13 15 15 14 14 14 14 13 13 13 13 11 11 11 11 11 11 11 11 [450491.765] VM(0/0/0) CDTC pBuf[35]: 15 15 15 15 16 15 14 15 16 15 14 14 14 14 14 13 15 15 14 14 14 14 13 13 13 13 11 11 11 11 11 11 11 11 11 [450491.775] VM(0/0/0) CDTC pBuf[35]: 16 16 16 16 15 14 15 16 15 14 14 14 14 14 13 15 15 14 14 14 14 13 13 13 13 11 11 11 11 11 11 11 11 11 11 [450492.365] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 14 15 14 14 14 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 [450492.375] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 15 14 14 14 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 [450492.385] VM(0/0/0) CDTC pBuf[35]: 15 15 15 15 14 14 14 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 [450492.395] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 14 14 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 [450492.405] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 14 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 [450492.415] VM(0/0/0) CDTC pBuf[35]: 14 14 14 14 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 [450492.425] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 [450492.435] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 11 [450492.445] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 11 11 [450492.455] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 11 11 11 [450492.465] VM(0/0/0) CDTC pBuf[35]: 12 12 12 12 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 11 11 11 11 [450492.475] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 [450492.485] VM(0/0/0) CDTC pBuf[35]: 13 13 13 13 13 12 12 11 12 12 12 13 12 14 13 13 13 12 12 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 Received SIP PDU from ( 213.170.81.135:5060 ) BYE sip:1722105@192.168.17.111 SIP/2.0 Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK19115e85;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111>;tag=ac600f8da4 Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 103 BYE User-Agent: VoipNow PBX Max-Forwards: 70 Content-Length: 0 671 <SIP 107> : Receive BYE Request Sending SIP PDU to ( 213.170.81.135:5060 ) from 5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 213.170.81.135:5060;branch=z9hG4bK19115e85;rport From: "101" <sip:101@213.170.81.135>;tag=as3eaf4bf9 To: <sip:1722105@192.168.17.111>;tag=ac600f8da4 Call-ID: 68af63e1620e40db0a9b20db728f2f9e@213.170.81.135 CSeq: 103 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 672 <SIP 107> : ReleaseWithNothing [450493.610] RTA(0/0/0) Rx RS_CLOSE_REQ callId=107 ssId=1 dir=reve [450493.610] RTA(0/0/0) Rx RS_CLOSE_REQ callId=107 ssId=1 dir=forw [450493.615] RTA(0/0/0) close Media socket [450493.615] RTA(0/0/0) close RTCP socket 673 <Call 107> : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0) 674 <CEP 000000> : DisconnectCall at Busy 675 <CEP 000000> : StopSignal [450493.615] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0 DTMF_STOP [450493.615] VM(0/0/0) play mute 676 <CEP 000000> : Disconnect (0) [450493.615] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0) [450493.615] VM(0/0/0) vopp idle [450493.615] VM(0/0/0) FXO OnHook [450493.620] VM(0/0/0) Tx DISCONN_CNF 677 <NetEP 107> : Call TO <101> terminated reason(Remote:CallClear) [450493.625] VM(0/0/0) Rx FXO Ring Actv [450493.625] VM(0/0/0) Tx RING_IND 678 <CEP 000000> : Disconnected(16) at Disconnecting 679 <CEP 000000> : Call Received [450493.660] VM(0/0/0) Rx FXO Ring Idle [450493.660] VM(0/0/0) Rx FXO Ring Ignore [450493.660] VM(0/0/0) Tx DISCONN_CNF 680 <CEP 000000> : Disconnected(16) at Busy conf Enter configuration commands, one per line. End with CNTL/Z AP200(config)# no deb? debug-port AP200(config)# no deb AP200(config)# x Invalid command (x) AP200(config)# ex AP200# ex [TELNET] INFO: DISCONNECTED * * DISCONNECT * time 16:00:39 *

Автор:	Николаев Игорь [ 26 май 2009, 13:23 ]
Заголовок сообщения:	Дополнение
когда игрался на fxo с gain-ми при h245-alphanumeric dtmf проходил лучше, но все равно неудовлитворительно, а rtl - тоже самое возможно есть какие-то апосы для стран снг со спец dtmf? или опять же я что-то не учел, какую-то настройку спрятанную?

Автор:	Geniu$$ [ 26 май 2009, 13:23 ]
Заголовок сообщения:
По этому дебагу видно, что все цифры пришли на шлюз и проигрываются в FXO.

Автор:	Николаев Игорь [ 26 май 2009, 13:32 ]
Заголовок сообщения:	Можно ещё погрешить на нерабочий адпак, но у меня их 2
Можно ещё погрешить на нерабочий адпак, но у меня их 2 и на обоих тоже самое Занято он у меня тоже не сразу распознавал. Только после конфигурации шаблона сигнала "занято" он начал его понимать. Возможно для генерации dtmf можно задать какие-то шаблоны, и если да то как и какие?

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