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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:

Спасибо за любую помощь.

Добрый день. Пришлите deb voip call, deb voip sip, deb rta ipc.

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 1722*105
port 0/0
user-password 53244183
!
dial-peer voice 1 pots
destination-pattern 1722*107
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

От куда идёт передача из IP в "город" или из "города" в IP?

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

Да. покажите дебаг

deb voip call
deb rta ipc
deb voip sip
conf t
deb

Включите dtmf-relay rtp-2833

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:1722*105@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:1722*105@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:1722*105@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:1722*107@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:1722*107@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:1722*107@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:1722*105@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:1722*105@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:1722*105@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(1722*105) callingParty(101)
657 <Call 107> : MatchedPerfect
658 <Call 107> : MatchAllProcess After Sorted
<0> id(0) dest(1722*105) prefer(0) selected(44)
659 <Call 107> : Initiate callee with dial-peer(1722*105) 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:1722*105@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:1722*105@192.168.17.111
Content-Type: application/sdp
Content-Length: 216

v=0
o=1722*105 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:1722*105@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:1722*105@192.168.17.111
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 216

v=0
o=1722*105 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:1722*105@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:1722*105@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:1722*105@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:1722*105@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:1722*105@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
***

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?

По этому дебагу видно, что все цифры пришли на шлюз и проигрываются в FXO.

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?