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Такая проблема. При звонках с аналогового телефона на addpac попадаю на disa и набираю донабор внутр. тел. сотрудника. Все работает. Все тоже самое но с цифрового телефона нет (На disa попадает, а дальше никак). В чем может быть проблема? Чем отличает dtfm аналог. телеф. от цифрового. С input gain, output gain, high-dtmf-gain, low-dtmf-gain значения менял, не реагирует

Выложите дебаги RTA IPC, voip call, voip sip. Для примера можно с обоих телефонов, при удачном и неудачном чтоб легче было определить разницу.

DEBUG С АНАЛОГОВОГО ТЕЛЕФОНА (донабор работает)
Received SIP PDU from ( 10.204.48.254:5060 )
INVITE sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00007d62
Max-Forwards: 70
To: sip:316@10.204.48.254
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=3754
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 1 INVITE
Contact: sip:10.204.48.254:5060
Supported: timer
Session-Expires: 90;refresher=uac
Min-SE: 90
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 1 1 IN IP4 10.204.48.253
s=-
c=IN IP4 10.204.48.253
t=0 0
m=audio 12226 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20


Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00007d62
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=3754
To: sip:316@10.204.48.254
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


1 <Call 22> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(470) ****
2 <SIP 22> : Receive INVITE Request

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00007d62
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=3754
To: sip:316@10.204.48.254
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 1 INVITE
Min-SE: 1800
User-Agent: AddPac SIP Gateway
Content-Length: 0


3 <SIP 22> : ReleaseWithNothing
4 <Call 22> : Terminated from(fffffffe) this(Remote:NoPermission) be
fore(NULL) forced(0) time(470)
5 <NetEP 22> : Call TO <> terminated reason(Remote:NoPermission)

Received SIP PDU from ( 10.204.48.254:5060 )
ACK sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00007d62
Max-Forwards: 70
To: sip:316@10.204.48.254
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=3754
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 1 ACK
Content-Length: 0


6 <SIP 22> : Receive ACK Request

Received SIP PDU from ( 10.204.48.254:5060 )
INVITE sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00005dda
Max-Forwards: 70
To: sip:316@10.204.48.254
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 2 INVITE
Contact: sip:10.204.48.254:5060
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 1800
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 1 1 IN IP4 10.204.48.253
s=-
c=IN IP4 10.204.48.253
t=0 0
m=audio 12226 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20


Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00005dda
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
To: sip:316@10.204.48.254
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


7 <Call 23> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(470) ****
8 <SIP 23> : Receive INVITE Request
9 <NetCon 23> : Found inbound voip peer by dest-pattern id(1000)
10 <Call 23> : From Net - calledParty(316) callingParty(400)
11 <Call 23> : MatchedPerfect
12 <Call 23> : MatchAllProcess After Sorted
<0> id(1) dest(316) prefer(0) selected(4)
13 <Call 23> : Initiate callee with dial-peer(316) status(CalleeDeter
minedAll) id(00000000-0000-0000-0000-000000000000)
14 <CEP 000100> : InitiateOutCall : calledNum(), callingNum(400), calle
rPort(ffffffff) type(FXS)
[479.420] RTA(0/1/0) Rx CC_RING_REQ [80 19 01 08 30 31 30 31 30 30 30 37 02 03 3
4 30 30 07 08 d2 e5 f1 f2 ee e2 fb e9 ] peerId(-1)
[479.420] VM(0/1/0) DaTime [L=8] 30 31 30 31 30 30 30 37
[479.420] VM(0/1/0) CgNumb [L=3] 34 30 30
[479.420] VM(0/1/0) CgName [L=8] d2 e5 f1 f2 ee e2 fb e9
[479.420] VM(0/1/0) Line Reverse
[479.420] VM(0/1/0) Start ring actv
[479.420] VM(0/1/0) SW to -72V
15 <CEP 000100> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(23)
[479.420] VM(0/1/0) set T38 enable by CCC
[479.420] VM(0/1/0) set T38 mode STD
[479.420] VM(0/1/0) Fax rate 9600
16 <SIP 23> : SetAlerting

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00005dda
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
To: sip:316@10.204.48.254;tag=d600f025a4
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 2 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:316@10.204.131.40
Content-Length: 0


[479.700] VM(0/1/0) vmOffHook
[479.760] VM(0/1/0) vmTmoOffHook
[479.760] VM(0/1/0) SW to -48V
[479.760] VM(0/1/0) Line Forward
[479.820] VM(0/1/0) vmTmoOffHook
[479.820] VM(0/1/0) Rx OffHook
[479.820] VP(0/1/0) open channel
[479.820] VP(0/1/0) attribute Fax enable, Modem disable
[479.820] VP(0/1/0) update Fax enable, Modem disable
[479.820] VM(0/1/0) Tx CONNECT_CNF
17 <Call 23> : Connected from(100)
[479.820] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[479.820] VM(0/1/0) VAD disable
[479.820] VP(0/1/0) update VAD 0
[479.820] VM(0/1/0) SID disable by CCC
18 <SIP 23> : SetConnected
19 <SIP 23> : Add Local Audio MediaFormat : 8

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00005dda
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
To: sip:316@10.204.48.254;tag=d600f025a4
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 2 INVITE
Supported: timer, replaces, early-session
Session-Expires: 1800;refresher=uac
User-Agent: AddPac SIP Gateway
Contact: sip:316@10.204.131.40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Require: timer
Content-Type: application/sdp
Content-Length: 225

v=0
o=316 471 471 IN IP4 10.204.131.40
s=AddPac Gateway SDP
c=IN IP4 10.204.131.40
t=0 0
m=audio 23038 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=sendrecv

[479.835] RTA(0/1/0) Rx RS_LISTEN_REQ callId=23 ssId=1 G711A
peer=10.204.48.253 mp=23038/23039 hp=12226/12227
[479.840] VM(0/1/0) vopp idle
[479.840] VP(0/1/0) close channel
[479.840] VM(0/1/0) start codec replace timer to G711A
[479.840] RTA(0/1/0) Rx RS_OPEN_REQ callId=23 ssId=1 G711A
peer=10.204.48.253 mp=23038/23039 hp=12226/12227
[479.840] VM(0/1/0) under codec replace to G711A
[479.840] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[479.840] VM(0/1/0) DTMF_RTP_RFC2833 enable
[479.840] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65

Received SIP PDU from ( 10.204.48.254:5060 )
ACK sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK000044b0
Max-Forwards: 70
To: sip:316@10.204.48.254;tag=d600f025a4
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 2 ACK
Content-Length: 0


20 <SIP 23> : ACK received
21 <SIP 23> : Receive ACK Request
22 <SIP 23> : Set Terminated Success for 2 INVITE
[479.895] VP(0/1/0) enable Fax, disable Modem
[479.895] VP(0/1/0) open channel
[479.895] VM(0/1/0) codec replaced to G711A
[479.895] VM(0/1/0) play mute
[479.895] VP(0/1/0) Tx IBS signal 2/0
[479.895] VP(0/1/0) Tx IBS dir 0
[479.960] VP(0/1/0) GeneralEvent IBS gen end
[481.585] VM(0/1/0) play digit single '1'
[481.585] VP(0/1/0) Tx IBS signal 1/1
[481.585] VP(0/1/0) Tx IBS dir 0
[481.685] RTA(0/1/0) RTP play DTMF stop
[481.685] VM(0/1/0) play DTMF stop
[481.685] VP(0/1/0) Tx IBS stop
[481.705] RTA(0/1/0) RTP play DTMF stop
[481.705] VM(0/1/0) play mute
[481.705] VP(0/1/0) Tx IBS signal 2/0
[481.705] VP(0/1/0) Tx IBS dir 0
[481.735] RTA(0/1/0) RTP play DTMF stop
[481.735] VM(0/1/0) play mute
[481.735] VP(0/1/0) Tx IBS signal 2/0
[481.735] VP(0/1/0) Tx IBS dir 0
[481.805] VP(0/1/0) GeneralEvent IBS gen end
[481.805] VM(0/1/0) play digit single '1'
[481.805] VP(0/1/0) Tx IBS signal 1/1
[481.805] VP(0/1/0) Tx IBS dir 0
[481.905] RTA(0/1/0) RTP play DTMF stop
[481.905] VM(0/1/0) play DTMF stop
[481.905] VP(0/1/0) Tx IBS stop
[481.925] RTA(0/1/0) RTP play DTMF stop
[481.925] VM(0/1/0) play mute
[481.925] VP(0/1/0) Tx IBS signal 2/0
[481.925] VP(0/1/0) Tx IBS dir 0
[481.955] RTA(0/1/0) RTP play DTMF stop
[481.955] VM(0/1/0) play mute
[481.955] VP(0/1/0) Tx IBS signal 2/0
[481.955] VP(0/1/0) Tx IBS dir 0
[482.025] VP(0/1/0) GeneralEvent IBS gen end
[482.125] VM(0/1/0) play digit single '3'
[482.125] VP(0/1/0) Tx IBS signal 1/3
[482.125] VP(0/1/0) Tx IBS dir 0
[482.245] RTA(0/1/0) RTP play DTMF stop
[482.245] VM(0/1/0) play DTMF stop
[482.245] VP(0/1/0) Tx IBS stop
[482.265] RTA(0/1/0) RTP play DTMF stop
[482.265] VM(0/1/0) play mute
[482.265] VP(0/1/0) Tx IBS signal 2/0
[482.265] VP(0/1/0) Tx IBS dir 0
[482.295] RTA(0/1/0) RTP play DTMF stop
[482.295] VM(0/1/0) play mute
[482.295] VP(0/1/0) Tx IBS signal 2/0
[482.295] VP(0/1/0) Tx IBS dir 0
[482.365] VP(0/1/0) GeneralEvent IBS gen end
23 <SIP 22> : Set Terminated Success for 1 INVITE

Received SIP PDU from ( 10.204.48.254:5060 )
BYE sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00001ddb
Max-Forwards: 70
To: sip:316@10.204.48.254;tag=d600f025a4
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 3 BYE
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Length: 0


24 <SIP 23> : Receive BYE Request

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00001ddb
From: "Òåñòîâûé" <sip:400@10.204.48.254>;tag=18872
To: sip:316@10.204.48.254;tag=d600f025a4
Call-ID: 00006df9-35218c5e1f5d10009b190080f0a4b648@10.204.48.254
CSeq: 3 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0


25 <SIP 23> : ReleaseWithNothing
[507.180] RTA(0/1/0) Rx RS_CLOSE_REQ callId=23 ssId=1 dir=all
[507.180] RTA(0/1/0) close Media socket
[507.180] RTA(0/1/0) close RTCP socket
26 <Call 23> : Terminated from(fffffffe) this(Remote:CallClear) befor
e(NULL) forced(0) time(498)
27 <CEP 000100> : DisconnectCall at Busy
28 <CEP 000100> : StopSignal
[507.185] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[507.185] VM(0/1/0) play mute
[507.185] VP(0/1/0) Tx IBS signal 2/0
[507.185] VP(0/1/0) Tx IBS dir 0
29 <CEP 000100> : Disconnect (0)
[507.185] RTA(0/1/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[507.185] VM(0/1/0) play Reorder tone
[507.185] VP(0/1/0) Tx IBS signal 6/3
[507.185] VP(0/1/0) Tx IBS dir 0
30 <NetEP 23> : Call FROM <Òåñòîâûé> terminated reason(Remote:CallClea
r)
[509.760] VM(0/1/0) vmOnHook
[509.810] VM(0/1/0) vmTmoOnHook
[509.860] VM(0/1/0) vmTmoOnHook
[509.910] VM(0/1/0) vmTmoOnHook
[509.960] VM(0/1/0) vmTmoOnHook
[510.010] VM(0/1/0) vmTmoOnHook
[510.060] VM(0/1/0) vmTmoOnHook
[510.110] VM(0/1/0) vmTmoOnHook
[510.160] VM(0/1/0) vmTmoOnHook
[510.210] VM(0/1/0) vmTmoOnHook
[510.260] VM(0/1/0) vmTmoOnHook
[510.310] VM(0/1/0) vmTmoOnHook
[510.360] VM(0/1/0) vmTmoOnHook
[510.410] VM(0/1/0) vmTmoOnHook
[510.460] VM(0/1/0) vmTmoOnHook
[510.460] VM(0/1/0) Rx OnHook
[510.460] VM(0/1/0) vopp idle
[510.460] VP(0/1/0) close channel
[510.460] VM(0/1/0) Tx DISCONN_CNF
31 <CEP 000100> : Disconnected(16) at Disconnecting
32 <SIP 23> : Set Terminated Success for 3 BYE
33 <Time 0> : SIP_TREGISTER timer timeout.
34 <SIP 0> : Adding authentication information
35 <SIP 15> : Send REGISTER Request

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
REGISTER sip:10.204.48.254 SIP/2.0
Via: SIP/2.0/UDP 10.204.131.40:5060;branch=z9hG4bK0900f100a415
From: <sip:315@10.204.48.254>;tag=0900f100a4
To: sip:315@10.204.48.254
Call-ID: 09000000-e91e-f167-8000-0002a406e75e@10.204.131.40
CSeq: 15 REGISTER
Date: Thu, 01 Jan 2010 00:10:51 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="....", realm="Registered Users", nonce="76ecd9b3
67ce9c3973e7ce9c3973e7ce", uri="sip:10.204.48.254", response="2f658f6af09496b9fb
9683b097b8031f", algorithm=MD5
Contact: <sip:315@10.204.131.40>;expires=302
Expires: 302
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 10.204.48.254:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.204.131.40:5060;branch=z9hG4bK0900f100a415
To: sip:315@10.204.48.254;tag=30349
From: sip:315@10.204.48.254;tag=0900f100a4
Call-ID: 09000000-e91e-f167-8000-0002a406e75e@10.204.131.40
CSeq: 15 REGISTER
Contact: sip:315@10.204.131.40;expires=301
Expires: 301
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Length: 0


36 <SIP 15> : Receive 200 OK
37 <SIP 15> : Transaction (15 REGISTER) completed
38 <SIP 0> : Adding authentication information
39 <SIP 16> : Send REGISTER Request

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
REGISTER sip:10.204.48.254 SIP/2.0
Via: SIP/2.0/UDP 10.204.131.40:5060;branch=z9hG4bKa700151ba416
From: <sip:316@10.204.48.254>;tag=a700151ba4
To: sip:316@10.204.48.254
Call-ID: a7000000-5fb7-15a7-801b-0002a406e75e@10.204.131.40
CSeq: 16 REGISTER
Date: Thu, 01 Jan 2010 00:10:51 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="....", realm="Registered Users", nonce="9c3973e6
cd9b366cd8b163c68d1b366c", uri="sip:10.204.48.254", response="79b7e4d83967e736e0
abd5c2abae1201", algorithm=MD5
Contact: <sip:316@10.204.131.40>;expires=302
Expires: 302
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 10.204.48.254:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.204.131.40:5060;branch=z9hG4bKa700151ba416
To: sip:316@10.204.48.254;tag=4079
From: sip:316@10.204.48.254;tag=a700151ba4
Call-ID: a7000000-5fb7-15a7-801b-0002a406e75e@10.204.131.40
CSeq: 16 REGISTER
Contact: sip:316@10.204.131.40;expires=302
Expires: 302
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Length: 0


40 <SIP 16> : Receive 200 OK
41 <SIP 16> : Transaction (16 REGISTER) completed
42 <SIP 15> : Set Terminated Success for 15 REGISTER
43 <SIP 16> : Set Terminated Success for 16 REGISTER



DEBUG С ЦИФРОВОГО ТЕЛЕФОНА (донабор не работает)
Received SIP PDU from ( 10.204.48.254:5060 )
INVITE sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK0000501f
Max-Forwards: 70
To: sip:316@10.204.48.254
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=20030
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 1 INVITE
Contact: sip:10.204.48.254:5060
Supported: timer
Session-Expires: 90;refresher=uac
Min-SE: 90
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 1 1 IN IP4 10.204.48.253
s=-
c=IN IP4 10.204.48.253
t=0 0
m=audio 12234 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20


Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK0000501f
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=20030
To: sip:316@10.204.48.254
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 1 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


12 <Call 24> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(903) ****
13 <SIP 24> : Receive INVITE Request

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK0000501f
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=20030
To: sip:316@10.204.48.254
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 1 INVITE
Min-SE: 1800
User-Agent: AddPac SIP Gateway
Content-Length: 0


14 <SIP 24> : ReleaseWithNothing
15 <Call 24> : Terminated from(fffffffe) this(Remote:NoPermission) be
fore(NULL) forced(0) time(903)
16 <NetEP 24> : Call TO <> terminated reason(Remote:NoPermission)

Received SIP PDU from ( 10.204.48.254:5060 )
ACK sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK0000501f
Max-Forwards: 70
To: sip:316@10.204.48.254
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=20030
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 1 ACK
Content-Length: 0


17 <SIP 24> : Receive ACK Request

Received SIP PDU from ( 10.204.48.254:5060 )
INVITE sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00003f87
Max-Forwards: 70
To: sip:316@10.204.48.254
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 INVITE
Contact: sip:10.204.48.254:5060
Supported: timer
Session-Expires: 1800;refresher=uac
Min-SE: 1800
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Type: application/sdp
Content-Length: 254

v=0
o=- 1 1 IN IP4 10.204.48.253
s=-
c=IN IP4 10.204.48.253
t=0 0
m=audio 12234 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20


Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00003f87
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
To: sip:316@10.204.48.254
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


18 <Call 25> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(903) ****
19 <SIP 25> : Receive INVITE Request
20 <NetCon 25> : Found inbound voip peer by dest-pattern id(1000)
21 <Call 25> : From Net - calledParty(316) callingParty(143)
22 <Call 25> : MatchedPerfect
23 <Call 25> : MatchAllProcess After Sorted
<0> id(1) dest(316) prefer(0) selected(5)
24 <Call 25> : Initiate callee with dial-peer(316) status(CalleeDeter
minedAll) id(00000000-0000-0000-0000-000000000000)
25 <CEP 000100> : InitiateOutCall : calledNum(), callingNum(143), calle
rPort(ffffffff) type(FXS)
[912.140] RTA(0/1/0) Rx CC_RING_REQ [80 1f 01 08 30 31 30 31 30 30 31 35 02 03 3
1 34 33 07 0e c1 ee f0 ee e4 f3 eb e8 ed 20 ca 2e c2 2e ] peerId(-1)
[912.140] VM(0/1/0) DaTime [L=8] 30 31 30 31 30 30 31 35
[912.140] VM(0/1/0) CgNumb [L=3] 31 34 33
[912.140] VM(0/1/0) CgName [L=14] c1 ee f0 ee e4 f3 eb e8 ed 20 ca 2e c2 2e
[912.140] VM(0/1/0) Line Reverse
[912.140] VM(0/1/0) Start ring actv
[912.140] VM(0/1/0) SW to -72V
26 <CEP 000100> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(25)
[912.140] VM(0/1/0) set T38 enable by CCC
[912.140] VM(0/1/0) set T38 mode STD
[912.140] VM(0/1/0) Fax rate 9600
27 <SIP 25> : SetAlerting

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00003f87
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
To: sip:316@10.204.48.254;tag=8700e527a4
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:316@10.204.131.40
Content-Length: 0


[912.420] VM(0/1/0) vmOffHook
[912.480] VM(0/1/0) vmTmoOffHook
[912.480] VM(0/1/0) SW to -48V
[912.480] VM(0/1/0) Line Forward
[912.540] VM(0/1/0) vmTmoOffHook
[912.540] VM(0/1/0) Rx OffHook
[912.540] VP(0/1/0) open channel
[912.540] VP(0/1/0) attribute Fax enable, Modem disable
[912.540] VP(0/1/0) update Fax enable, Modem disable
[912.540] VM(0/1/0) Tx CONNECT_CNF
28 <Call 25> : Connected from(100)
[912.540] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[912.540] VM(0/1/0) VAD disable
[912.540] VP(0/1/0) update VAD 0
[912.540] VM(0/1/0) SID disable by CCC
29 <SIP 25> : SetConnected
30 <SIP 25> : Add Local Audio MediaFormat : 8

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK00003f87
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
To: sip:316@10.204.48.254;tag=8700e527a4
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 INVITE
Supported: timer, replaces, early-session
Session-Expires: 1800;refresher=uac
User-Agent: AddPac SIP Gateway
Contact: sip:316@10.204.131.40
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Require: timer
Content-Type: application/sdp
Content-Length: 225

v=0
o=316 903 903 IN IP4 10.204.131.40
s=AddPac Gateway SDP
c=IN IP4 10.204.131.40
t=0 0
m=audio 23046 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=sendrecv

[912.555] RTA(0/1/0) Rx RS_LISTEN_REQ callId=25 ssId=1 G711A
peer=10.204.48.253 mp=23046/23047 hp=12234/12235
[912.560] VM(0/1/0) vopp idle
[912.560] VP(0/1/0) close channel
[912.560] VM(0/1/0) start codec replace timer to G711A
[912.560] RTA(0/1/0) Rx RS_OPEN_REQ callId=25 ssId=1 G711A
peer=10.204.48.253 mp=23046/23047 hp=12234/12235
[912.560] VM(0/1/0) under codec replace to G711A
[912.560] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[912.560] VM(0/1/0) DTMF_RTP_RFC2833 enable
[912.560] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
Rtp2833_DtmfPT TxPT=0x65 RxPT=0x65

Received SIP PDU from ( 10.204.48.254:5060 )
ACK sip:316@10.204.131.40 SIP/2.0
Via: SIP/2.0/UDP 10.204.48.254:5060;branch=z9hG4bK000034d9
Max-Forwards: 70
To: sip:316@10.204.48.254;tag=8700e527a4
From: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 ACK
Content-Length: 0


31 <SIP 25> : ACK received
32 <SIP 25> : Receive ACK Request
33 <SIP 25> : Set Terminated Success for 2 INVITE
[912.615] VP(0/1/0) enable Fax, disable Modem
[912.615] VP(0/1/0) open channel
[912.615] VM(0/1/0) codec replaced to G711A
[912.615] VM(0/1/0) play mute
[912.615] VP(0/1/0) Tx IBS signal 2/0
[912.615] VP(0/1/0) Tx IBS dir 0
[912.685] VP(0/1/0) GeneralEvent IBS gen end
34 <SIP 24> : Set Terminated Success for 1 INVITE
[931.760] VM(0/1/0) vmOnHook
[931.810] VM(0/1/0) vmTmoOnHook
[931.860] VM(0/1/0) vmTmoOnHook
[931.910] VM(0/1/0) vmTmoOnHook
[931.960] VM(0/1/0) vmTmoOnHook
[932.010] VM(0/1/0) vmTmoOnHook
[932.060] VM(0/1/0) vmTmoOnHook
[932.110] VM(0/1/0) vmTmoOnHook
[932.160] VM(0/1/0) vmTmoOnHook
[932.210] VM(0/1/0) vmTmoOnHook
[932.260] VM(0/1/0) vmTmoOnHook
[932.310] VM(0/1/0) vmTmoOnHook
[932.360] VM(0/1/0) vmTmoOnHook
[932.410] VM(0/1/0) vmTmoOnHook
[932.460] VM(0/1/0) vmTmoOnHook
[932.460] VM(0/1/0) Rx OnHook
[932.460] VM(0/1/0) vopp idle
[932.460] VP(0/1/0) close channel
[932.460] VM(0/1/0) Tx DISCONN_CNF
35 <CEP 000100> : Disconnected(16) at Busy
36 <Call 25> : Terminated from(100) this(Local:CallClear) before(NULL
) forced(0) time(923)
37 <SIP 25> : ReleaseWithBYE
38 <SIP 25> : Send BYE Request

Sending SIP PDU to ( 10.204.48.254:5060 ) from 5060
BYE sip:10.204.48.254 SIP/2.0
Via: SIP/2.0/UDP 10.204.131.40:5060;branch=z9hG4bK8700e527a42
From: sip:316@10.204.48.254;tag=8700e527a4
To: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 BYE
Date: Thu, 01 Jan 2010 00:15:23 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:316@10.204.131.40>
Contact: <sip:316@10.204.131.40>
Max-Forwards: 70


[932.465] RTA(0/1/0) Rx RS_CLOSE_REQ callId=25 ssId=1 dir=all
[932.470] RTA(0/1/0) close Media socket
[932.470] RTA(0/1/0) close RTCP socket
39 <NetEP 25> : Call FROM <Áîðîäóëèí Ê.Â.> terminated reason(Local:Cal
lClear)
40 <CEP 000100> : DisconnectCall at Idle

Received SIP PDU from ( 10.204.48.254:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.204.131.40:5060;branch=z9hG4bK8700e527a42
To: "Áîðîäóëèí Ê.Â." <sip:143@10.204.48.254>;tag=7002
From: sip:316@10.204.48.254;tag=8700e527a4
Call-ID: 000018d5-35218c5e46a710009b1a0080f0a4b648@10.204.48.254
CSeq: 2 BYE
Content-Length: 0


41 <SIP 25> : Receive 200 OK
42 <SIP 25> : Transaction (2 BYE) completed
43 <SIP 25> : Set Terminated Success for 2 BYE

У Вас в дебаге звонок идет не с телефонов, а на них(тоесть что они донабирают?). И где находиться сама Disa, стоит перед станциией(какойто АТС-кой)?

1. на АТС TDE200 прописаны все внутр. номера, + sip номера
2 addpac присвоили один из sip номер, прописанных на TDE200
3 addpac поставили на KX-TEM824, на этой станции стоит disa.
4. Звоним с TDE200 звонок попадает на addpac потом он попадает на DISA ждет донабора внутр. номера которые прописаны на станции TEM824

пример: на TDE200 прописан внутр. 110, на TEM824 внут 105. Звоню с 110 на (300) addpac (на addpac предположим номер 300 прописан), срабатывает disa и ждет донабора с аналого. пред. набираю 105 или любой которые из внутр. прописанных на TEM824

1)Да, теперь все понятно (красиво расписали )
2)Судя по дебагам вы донабираете в первом случае с аналогового вроде телефон 113. Во втором случае ничего не идет(но диса отвечает). Как на меня так странная ситуация, до адпака ведь цифровой телефон то дозваниваеться легко, а почему при донаборе ничего неидет неясно. Может это какието спецификации самого телефона, там в настройках чето, другой телефон непробовали (цифровой)?

Да для теста набираю 113. Получается, когда звоню с цифрового, на disa попадает и disa ждет донабора, но какие бы кнопки не нажимал на цифровом телефоне, addpac или KX-TEM824 никак не реагирует. Аналоговый реагирует сразу. Когда позвонил с анал. нажимаю flash перек. на цифровой, канал держится все говорит Цифровой телефон какой сигнал передает, тоже dtfm? Disa отдельно проверял, звонил через disa без addpac, донабор работает. В организации как цифровые телефоны, так и аналоговые. Пробовал с разных цифр. тел. результат один. Может что-то в конфиге добавить?

version 8.41.086
!
hostname AP100B
!
username root password router administrator
!
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 10.204.131.40 255.255.255.0
ip nat outside
speed auto
no qos-control
!
interface FastEthernet0/1
ip address 192.168.10.1 255.255.255.0
ip nat inside
speed auto
no qos-control
!
ip route 0.0.0.0 0.0.0.0 10.204.131.1 10
!
access-list 100 permit ip 192.168.10.0 0.0.0.255 any
!
!
ip nat inside source list 100 interface FastEthernet0/0 overload
!
!
snmp name AP100B_G2
!
!
ftp server
http server
!
!
dns name-server 10.204.40.30
dns name-server 10.0.101.9
logging event 0-emergency
!
!
!
!
! 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
busyout monitor gatekeeper
busyout monitor sip-server
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 315
port 0/0
call-waiting
user-password ....
!
dial-peer voice 1 pots
destination-pattern 316
port 0/1
call-waiting
user-password ....
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
no sid
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.204.131.40
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g7231r63
codec preference 2 g729
codec preference 3 g711ulaw
codec preference 4 g711alaw
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-username ....
sip-password ....
sip-server 10.204.48.254
register e164
!
!
! MGCP configuration.
!
mgcp
vad
!
!
! Tones
!
!
!
!
line console
!
line vty
!
!
sms
quota 30
!
end
AP100B(config)#

Нужно подождать может специ чето и подскажут, хотя непонятно, если б чето нетак было с настройками цифровых то он бы наверное и до адпака недозванивался, а так же звонит, только проблема что недонабирает почемуто(да подебагу видно что вообще ниче неидет, такое может быть при нетональном наборе, адпак пульс непонимает, но они цифровые так что все ок, а может действительно достаточно какуйто галочку доставить и все будет ок(хотя незнаю что в адпаке моно поменять для цифрового, кроме как силу дтмф, но вы ж меняли ничего недает, так что сложновато сказать в чем проблема, может какраз чето в телефонах, там их както покрутит можно, хотя недогадываюсь что). А вообще действительно странно, и дебаг проходить точно так же для обоих ситуций только во второй недонабирает а так все идентично.

С input gain, output gain, high-dtmf-gain, low-dtmf-gain есть какие-нибудь рекомендуемые параметры, может не подобрал

Ну input output gain сдесь вроде как непричем, они для слышимости(звук, голос). А вот рекомендуемых high-dtmf-gain, low-dtmf-gain впринципе особо нету. Вообще восновном у всех все работает с настройками по умолчанию, а так подстраивают как могут, так что сложно сказать.