поставил в Inboudroutes
description : audiocodes.com
и звонки стали приниматься в назначенную группу..

а возмржны не правельные тональные посылки с FXO на ГАТС?

От телефона зависит, некоторые телефоны имеют сбитый тон. Занимаясь карточной платформой такое часто бывало.

звоню Zoipera там я думаю не должны тональности сбиваться..
подскажите что не так...
с города прием есть, а в город НЕТ...

-- Called PSTN/729745
-- SIP/PSTN-09984e40 is making progress passing it to SIP/100-099c4040
-- Got SIP response 500 "Server Internal Error" back from 192.168.0.7
-- SIP/PSTN-09984e40 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/100-099c4040", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/100-099c4040", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/100-099c4040", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [729745@from-internal:5] Macro("SIP/100-099c4040", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/100-099c4040", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/100-099c4040> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/100-099c4040", "pls-try-call-later|noanswer") in new stack
-- <SIP/100-099c4040> Playing 'pls-try-call-later' (language 'en')

sip set debug peer PSTN

trixbox1*CLI> sip set debug peer PSTN
SIP Debugging Enabled for IP: 192.168.0.7:5060
-- Executing [222222@from-internal:1] Macro("SIP/100-08e1e6b8", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/100-08e1e6b8", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/100-08e1e6b8", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/100-08e1e6b8", "1|Set|REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/100-08e1e6b8", "AMPUSER=100") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/100-08e1e6b8", "AMPUSERCIDNAME=100") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/100-08e1e6b8", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/100-08e1e6b8", "AMPUSERCID=100") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/100-08e1e6b8", "CALLERID(all)="100" <100>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/100-08e1e6b8", "REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/100-08e1e6b8", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/100-08e1e6b8", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/100-08e1e6b8", "Using CallerID "100" <100>") in new stack
-- Executing [222222@from-internal:2] Set("SIP/100-08e1e6b8", "_NODEST=") in new stack
-- Executing [222222@from-internal:3] Macro("SIP/100-08e1e6b8", "record-enable|100|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/100-08e1e6b8", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/100-08e1e6b8", "recordingcheck|20090430-131704|1241057824.10") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090430-131704|1241057824.10: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/100-08e1e6b8", "") in new stack
-- Executing [222222@from-internal:4] Macro("SIP/100-08e1e6b8", "dialout-trunk|2|222222||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/100-08e1e6b8", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/100-08e1e6b8", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/100-08e1e6b8", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/100-08e1e6b8", "DIAL_NUMBER=222222") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/100-08e1e6b8", "DIAL_TRUNK_OPTIONS=tTr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/100-08e1e6b8", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/100-08e1e6b8", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/100-08e1e6b8", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/100-08e1e6b8", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/100-08e1e6b8", "outbound-callerid|2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/100-08e1e6b8", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/100-08e1e6b8", "0|Set|REALCALLERIDNUM=100") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/100-08e1e6b8", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/100-08e1e6b8", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/100-08e1e6b8", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/100-08e1e6b8", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/100-08e1e6b8", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/100-08e1e6b8", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/100-08e1e6b8", "1?exit") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/100-08e1e6b8", "") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/100-08e1e6b8", "0|AGI|fixlocalprefix") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/100-08e1e6b8", "OUTNUM=222222") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/100-08e1e6b8", "custom=SIP/PSTN") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-08e1e6b8", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-08e1e6b8", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-08e1e6b8", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-08e1e6b8", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-08e1e6b8", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-08e1e6b8", "SIP/PSTN/222222|300|") in new stack
Video is at 192.168.0.1 port 11618
Audio is at 192.168.0.1 port 14042
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x200000 (h264) to SDP
Reliably Transmitting (no NAT) to 192.168.0.7:5060:
INVITE sip:222222@192.168.0.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0819c337;rport
From: "100" <sip:100@192.168.0.1>;tag=as1ccbeba4
To: <sip:222222@192.168.0.7>
Contact: <sip:100@192.168.0.1>
Call-ID: 515e8545278a3951335fa09221531452@192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 30 Apr 2009 02:17:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 2756 2756 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
b=CT:384
t=0 0
m=audio 14042 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 11618 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
-- Called PSTN/222222
trixbox1*CLI>
<--- SIP read from 192.168.0.7:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0819c337;rport
From: "100" <sip:100@192.168.0.1>;tag=as1ccbeba4
To: <sip:222222@192.168.0.7>;tag=1c215618136
Call-ID: 515e8545278a3951335fa09221531452@192.168.0.1
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXO/v.5.40A.003.001
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 192.168.0.7:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0819c337;rport
From: "100" <sip:100@192.168.0.1>;tag=as1ccbeba4
To: <sip:222222@192.168.0.7>;tag=1c215618136
Call-ID: 515e8545278a3951335fa09221531452@192.168.0.1
CSeq: 102 INVITE
Contact: <sip:729748@192.168.0.7>
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXO/v.5.40A.003.001
Content-Type: application/sdp
Content-Length: 364

v=0
o=AudiocodesGW 215648537 215648421 IN IP4 192.168.0.7
s=Phone-Call
c=IN IP4 192.168.0.7
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
a=rtcp:6001 IN IP4 192.168.0.7
m=video 0 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42000A; packetization-mode=0
a=sendrecv

<------------->
--- (12 headers 16 lines) ---
Found RTP audio format 0
Found RTP video format 34
Found RTP video format 99
Peer audio RTP is at port 192.168.0.7:6000
Found audio description format PCMU for ID 0
Found video description format H263 for ID 34
Found video description format H264 for ID 99
Capabilities: us - 0x28000c (ulaw|alaw|h263|h264), peer - audio=0x280004 (ulaw|h263|h264)/video=0x280000 (h263|h264), combined - 0x280004 (ulaw|h263|h264)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.7:6000
-- SIP/PSTN-08e63988 is making progress passing it to SIP/100-08e1e6b8
trixbox1*CLI>
<--- SIP read from 192.168.0.7:5060 --->
SIP/2.0 500 Server Internal Error
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0819c337;rport
From: "100" <sip:100@192.168.0.1>;tag=as1ccbeba4
To: <sip:222222@192.168.0.7>;tag=1c215618136
Call-ID: 515e8545278a3951335fa09221531452@192.168.0.1
CSeq: 102 INVITE
Supported: em,timer,replaces,path,early-session,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-114 FXO/v.5.40A.003.001
Reason: Q.850 ;cause=111
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
-- Got SIP response 500 "Server Internal Error" back from 192.168.0.7
Transmitting (no NAT) to 192.168.0.7:5060:
ACK sip:222222@192.168.0.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK0819c337;rport
From: "100" <sip:100@192.168.0.1>;tag=as1ccbeba4
To: <sip:222222@192.168.0.7>;tag=1c215618136
Contact: <sip:100@192.168.0.1>
Call-ID: 515e8545278a3951335fa09221531452@192.168.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/PSTN-08e63988 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] Goto("SIP/100-08e1e6b8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/100-08e1e6b8", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
Really destroying SIP dialog '515e8545278a3951335fa09221531452@192.168.0.1' Method: INVITE
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/100-08e1e6b8", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [222222@from-internal:5] Macro("SIP/100-08e1e6b8", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] Playback("SIP/100-08e1e6b8", "all-circuits-busy-now|noanswer") in new stack
-- <SIP/100-08e1e6b8> Playing 'all-circuits-busy-now' (language 'en')
-- Executing [s@macro-outisbusy:2] Playback("SIP/100-08e1e6b8", "pls-try-call-later|noanswer") in new stack
-- <SIP/100-08e1e6b8> Playing 'pls-try-call-later' (language 'en')
-- Executing [s@macro-outisbusy:3] Macro("SIP/100-08e1e6b8", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/100-08e1e6b8", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/100-08e1e6b8", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/100-08e1e6b8", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/100-08e1e6b8", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/100-08e1e6b8", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/100-08e1e6b8", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-08e1e6b8' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-08e1e6b8' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-08e1e6b8'

1. у аудиокодес есть свои прекрасные логи доступные через веб-морду.
если она вас посылает - логично на ней и смотреть почему. Там может быть вообще что угодно - не настроены группы исходящих, что угодно...

2. вам вообще зачем inband dtmf использовать? Настройте rfc2833 везде.

IP to Tel Calls Count

Number of Attempted Calls 2
Number of Established Calls 0
Percentage of Successful Calls(ASR) 0.000000
Number of Calls Terminated due to a Busy Line 0
Number of Calls Terminated due to No Answer 0
Number of Calls Terminated due to Forward 0
Number of Failed Calls due to No Route 0
Number of Failed Calls due to No Matched Capabilities 0
Number of Failed Calls due to No Resources 0
Number of Failed Calls due to Other Failures 2
Average Call Duration(ACD)[sec] 0
Attempted Fax Calls Counter 0
Successful Fax Calls Counter 0

Tel to IP Calls Count

Number of Attempted Calls 1
Number of Established Calls 1
Percentage of Successful Calls(ASR) 100.000000
Number of Calls Terminated due to a Busy Line 0
Number of Calls Terminated due to No Answer 0
Number of Calls Terminated due to Forward 0
Number of Failed Calls due to No Route 0
Number of Failed Calls due to No Matched Capabilities 0
Number of Failed Calls due to No Resources 0
Number of Failed Calls due to Other Failures 0
Average Call Duration(ACD)[sec] 176
Attempted Fax Calls Counter 0
Successful Fax Calls Counter 0

Call Routing Status

Current Call-Routing Method Proxy/GK
Current Proxy * IP (* IP)
Current Proxy State OK

Смущает вот это
Call Progress Tones File Name: usa_tones_12.dat

вот лог c putty

/>NOTIC:( lgr_flow)(1579 ) ---- Incoming SIP Message from 192.168.0.1:5060 ----

NOTIC:INVITNOTIC:( lgr_psbrdex)(1634 ) InsertBoardEvent- event 44 inserted channel 0

NOTIC:( lgr_fNOTIC:( lgr_psbrdex)(1670 ) InsertBoardEvent- event 116 inserted channel 0

NOTIC:( lgr_flow)(1671 ) #0:GUARD_TIME_TIMER_EXPIRED_EV

NOTIC:( lgr_flow)(1672 ) | #0:GUARD_TIME_TIMER_EXPIRED_EV

NOTIC:( lgr_psbrdif)(1673 ) #0:cpDigitMapHndlr_Stop - Stoped (0)

NOTIC:( lgr_psbrdif)(1674 ) #0:CloseChannel: ChannelNum=0

NOTIC:( lgr_psbrdif)(1675 ) Open channel: IsVoiceOn: 1, IsT38On: 1, IsVbdOn: 0, IsVideoOn: 0

NOTIC:( lgr_psbrdif)(1676 ) #0:OpenChannel:on Trunk -1 BChannel:0 CID=0 with VoiceCoder: g711Alaw64k20 VbdCoder: InvalidCoder255 DetectorSide: 0 FaxModemDet NO_FAX_MODEM_DETECTED

NOTIC:( lgr_psbrdif)(1677 ) #0:OpenChannel VoiceVolume= 0, DTMFVolume = -31, InputGain = 0, RTPRedundancyDepth = 0 FlashHookPeriod = 700


NOTIC:( lgr_psbrdif)(1678 ) OpenChannel, CoderType = 0, Interval = 3, M = 1


NOTIC:( lgr_psbrdif)(1679 ) #0:FAXTransportType = 1

NOTIC:( lgr_psbrdif)(1680 ) #0:ConfigFaxModemChannelParams NSEMode=0, CNGDetMode=0, FAXTranType=1, VxxTranType=2, VoiceVol= 0, DTMFVol=-31, InGain=0, RTPRedDepth=0, ECE=1, SCE=1, ECNlpMode=0, DJBufMinDelay=10, DJBufOptFac=10)

NOTIC:( lgr_psbrdif)(1681 ) Detectors: Amd:0, Ans:0 En:0 IBScmd:0xa1

NOTIC:( lgr_psbrdif)(1682 ) #0:cpDigitMapHndlr_Stop - Stoped (0)

NOTIC:( lgr_flow)(1683 ) | | TransactionUserMngr::ReturnSIPCall - #10

NOTIC:( sip_stack)(1684 ) SIPCall(#10) changes state from Disconnected to Idle