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Это реализуемо на этом оборудовании, кто нибудь так делал ?

Реализуемо.
Какое время flash на телефоне?
Возможно шлюз думает, что это уже не flash, а просто трубку положили!
На шлюзе по умолчанию:
tohdt (On Hook Detect Time) = 700 msec.

Покажите лог во время звонка
deb voip call
deb rta ipc
deb voip sip

Это время можно переставаить ?
на ком из шлюзов снимать дебаг FXO или FXS?

AP1100F# debug voip call
AP1100F# deb rta ipc
AP1100F# deb voip sip
AP1100F#
Received SIP PDU from ( 10.100.0.37:5060 )
INVITE sip:189@10.100.0.39 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.37:5060;branch=z9hG4bK6b4b3438a424
From: <sip:10.100.0.37>;tag=6b4b3438a4
To: <sip:189@10.100.0.39>
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Tue, 09 Feb 2010 16:29:31 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:10.100.0.37>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 252
Max-Forwards: 70

v=0
o=- 1265732971 1265732971 IN IP4 10.100.0.37
s=AddPac Gateway SDP
c=IN IP4 10.100.0.37
t=1265732971 0
m=audio 23042 RTP/AVP 18 4 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Sending SIP PDU to ( 10.100.0.37:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.100.0.37:5060;branch=z9hG4bK6b4b3438a424
From: <sip:10.100.0.37>;tag=6b4b3438a4
To: <sip:189@10.100.0.39>
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0


1 <Call 29> : ****************** Call Created status(InitiatedByNet) *******************
2 <SIP 29> : Receive INVITE Request
3 <NetCon 29> : Found inbound voip peer by IP address id(100)
4 <Call 29> : From Net - calledParty(189) callingParty()
5 <Call 29> : MatchedPerfect
6 <Call 29> : MatchAllProcess After Sorted
<0> id(1) dest(189) prefer(0) selected(18)
7 <Call 29> : Initiate callee with dial-peer(189) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
8 <CEP 000000> : InitiateOutCall : calledNum(), callingNum(), callerPort(ffffffff) type(FXS)
[4312.005] RTA(0/0/0) Rx CC_RING_REQ [80 1a 01 08 30 32 30 39 31 36 30 30 04 01 4f 07 0b 31 30 2e 31 30 30 2e 30 2e 33 37 ] peerId(-1)
[4312.005] VM(0/0/0) DaTime [L=8] 30 32 30 39 31 36 30 30
[4312.005] VM(0/0/0) CgNoNu [L=1] 4f
[4312.005] VM(0/0/0) CgName [L=11] 31 30 2e 31 30 30 2e 30 2e 33 37
[4312.010] VM(0/0/0) Line Reverse
[4312.010] VM(0/0/0) Start ring actv
[4312.010] VM(0/0/0) SW to -72V
9 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(29)
[4312.010] VM(0/0/0) set T38 mode STD
[4312.010] VM(0/0/0) Fax rate 9600
10 <SIP 29> : SetAlerting

Sending SIP PDU to ( 10.100.0.37:5060 ) from 5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.100.0.37:5060;branch=z9hG4bK6b4b3438a424
From: <sip:10.100.0.37>;tag=6b4b3438a4
To: <sip:189@10.100.0.39>;tag=b34bf424a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 INVITE
User-Agent: AddPac SIP Gateway
Contact: sip:189@10.100.0.39
Content-Length: 0


[4313.010] VM(0/0/0) Gen ring idle
[4313.435] VM(0/0/0) vmOffHook
[4313.495] VM(0/0/0) vmTmoOffHook
[4313.495] VM(0/0/0) SW to -48V
[4313.495] VM(0/0/0) Line Forward
[4313.555] VM(0/0/0) vmTmoOffHook
[4313.555] VM(0/0/0) Rx OffHook
[4313.555] VM(0/0/0) vopp enable
[4313.555] VM(0/0/0) Fax enable
[4313.555] VM(0/0/0) play mute
[4313.555] VM(0/0/0) Tx CONNECT_CNF
11 <Call 29> : Connected from(0)
[4313.555] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[4313.555] VM(0/0/0) VAD disable
[4313.555] VM(0/0/0) SID enable by CCC
12 <SIP 29> : SetConnected
13 <SIP 29> : Add Local Audio MediaFormat : 18

Sending SIP PDU to ( 10.100.0.37:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.37:5060;branch=z9hG4bK6b4b3438a424
From: <sip:10.100.0.37>;tag=6b4b3438a4
To: <sip:189@10.100.0.39>;tag=b34bf424a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:189@10.100.0.39
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 232

v=0
o=189 1265731252 1265731252 IN IP4 10.100.0.39
s=AddPac Gateway SDP
c=IN IP4 10.100.0.39
t=1265731252 0
m=audio 23040 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:20

[4313.595] RTA(0/0/0) Rx RS_LISTEN_REQ callId=29 ssId=1 G729A
peer=10.100.0.37 mp=23040/23041 hp=23042/23043
[4313.600] VM(0/0/0) vopp idle
[4313.600] VM(0/0/0) VoPP ready
[4313.600] VM(0/0/0) start codec replace timer to G729A
[4313.600] RTA(0/0/0) Rx RS_OPEN_REQ callId=29 ssId=1 G729A
peer=10.100.0.37 mp=23040/23041 hp=23042/23043
[4313.600] VM(0/0/0) under codec replace to G729A
[4313.600] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[4313.600] VM(0/0/0) DTMF_RTP_RFC2833 enable
[4313.600] VM(0/0/0) DTMF_RTP_RFC2833 TxPT=0x65, RxPT=0x65

Received SIP PDU from ( 10.100.0.37:5060 )
ACK sip:189@10.100.0.39 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.37:5060;branch=z9hG4bK6b4b3438a424
From: <sip:10.100.0.37>;tag=6b4b3438a4
To: <sip:189@10.100.0.39>;tag=b34bf424a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 ACK
Content-Length: 0
Max-Forwards: 70

14 <SIP 29> : ACK received
15 <SIP 29> : Receive ACK Request
16 <SIP 29> : Set Terminated Success for 24 INVITE
[4313.660] VM(0/0/0) vopp enable
[4313.660] VM(0/0/0) codec replaced to G729A
[4313.660] VM(0/0/0) Fax enable
[4313.660] VM(0/0/0) play mute
[4317.285] VM(0/0/0) vmOnHook
[4317.335] VM(0/0/0) vmTmoOnHook
[4317.355] VM(0/0/0) vmOffHook
[4317.415] VM(0/0/0) vmTmoOffHook
[4317.415] VM(0/0/0) Rx OffHook
[4317.415] VM(0/0/0) Tx FLASH_IND
17 <CEP 000000> : Hook Flashed
18 <Call 29> : HookFlashed from(0)
19 <SIP 29> : INFO send
20 <SIP 29> : Send INFO Request

Sending SIP PDU to ( 10.100.0.37:5060 ) from 5060
INFO sip:10.100.0.37 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.39:5060;branch=z9hG4bKb34bf424a423
From: <sip:189@10.100.0.39>;tag=b34bf424a4
To: <sip:10.100.0.37>;tag=6b4b3438a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 23 INFO
Date: Tue, 09 Feb 2010 16:00:56 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:189@10.100.0.39>
Content-Type: audio/telephone-event
Content-Length: 4
Max-Forwards: 70



Received SIP PDU from ( 10.100.0.37:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.39:5060;branch=z9hG4bKb34bf424a423
From: <sip:189@10.100.0.39>;tag=b34bf424a4
To: <sip:10.100.0.37>;tag=6b4b3438a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 23 INFO
User-Agent: AddPac SIP Gateway
Content-Length: 0

21 <SIP 29> : Receive 200 OK
22 <SIP 29> : Transaction (23 INFO) completed
23 <SIP 29> : Check Event Relation
24 <SIP 29> : Set Terminated Success for 23 INFO
[4322.605] VM(0/0/0) vmOnHook
[4322.655] VM(0/0/0) vmTmoOnHook
[4322.705] VM(0/0/0) vmTmoOnHook
[4322.755] VM(0/0/0) vmTmoOnHook
[4322.805] VM(0/0/0) vmTmoOnHook
[4322.855] VM(0/0/0) vmTmoOnHook
[4322.905] VM(0/0/0) vmTmoOnHook
[4322.955] VM(0/0/0) vmTmoOnHook
[4323.005] VM(0/0/0) vmTmoOnHook
[4323.055] VM(0/0/0) vmTmoOnHook
[4323.105] VM(0/0/0) vmTmoOnHook
[4323.155] VM(0/0/0) vmTmoOnHook
[4323.205] VM(0/0/0) vmTmoOnHook
[4323.255] VM(0/0/0) vmTmoOnHook
[4323.305] VM(0/0/0) vmTmoOnHook
[4323.305] VM(0/0/0) Rx OnHook
[4323.305] VM(0/0/0) vopp idle
[4323.305] VM(0/0/0) VoPP ready
[4323.305] VM(0/0/0) Tx DISCONN_CNF
25 <CEP 000000> : Disconnected(16) at Busy
26 <Call 29> : Terminated from(0) this(Local:CallClear) before(NULL) forced(0)
27 <SIP 29> : ReleaseWithBYE
28 <SIP 29> : Send BYE Request

Sending SIP PDU to ( 10.100.0.37:5060 ) from 5060
BYE sip:10.100.0.37 SIP/2.0
Via: SIP/2.0/UDP 10.100.0.39:5060;branch=z9hG4bKb34bf424a424
From: <sip:189@10.100.0.39>;tag=b34bf424a4
To: <sip:10.100.0.37>;tag=6b4b3438a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 BYE
Date: Tue, 09 Feb 2010 16:01:02 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:189@10.100.0.39>
Content-Length: 0
Max-Forwards: 70


[4323.325] RTA(0/0/0) Rx RS_CLOSE_REQ callId=29 ssId=1 dir=reve
[4323.325] RTA(0/0/0) Rx RS_CLOSE_REQ callId=29 ssId=1 dir=forw
[4323.325] RTA(0/0/0) close Media socket
[4323.325] RTA(0/0/0) close RTCP socket
29 <NetEP 29> : Call TO <10.100.0.37> terminated reason(Local:CallClear)
30 <CEP 000000> : DisconnectCall at Idle

Received SIP PDU from ( 10.100.0.37:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.100.0.39:5060;branch=z9hG4bKb34bf424a424
From: <sip:189@10.100.0.39>;tag=b34bf424a4
To: <sip:10.100.0.37>;tag=6b4b3438a4
Call-ID: 6b8d714b-eb48-3429-8038-0002a4040e48@10.100.0.37
CSeq: 24 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0

31 <SIP 29> : Receive 200 OK
32 <SIP 29> : Transaction (24 BYE) completed
33 <SIP 29> : Set Terminated Success for 24 BYE

Можно ещё раз последние конфиги.
попробуйте изменить dtmf на
dtmf-relay dual-mode
везде в воипных пирах.
Чувствую придется пинать разработчиков.

AP1005# sh run
!
version 8.30U
!
hostname AP1005
!
!
no bridge spanning-tree
!
!
no ip-share enable
!
interface ether0.0
ip address 10.100.0.37 255.255.255.0
!
interface ether1.0
no ip address
!
snmp name AP1005
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
route 0.0.0.0 0.0.0.0 10.100.0.1
!
logging event all
!
dnshost nameserver 213.170.64.33
dnshost nameserver 213.170.65.33
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 2
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
connection plar 189
ring detect-timeout 50
busyout action tone
no caller-id enable
!
!
! FXO
voice-port 0/1
connection plar 190
ring detect-timeout 50
busyout action tone
no caller-id enable
!
!
! FXO
voice-port 0/2
connection plar 191
ring detect-timeout 50
busyout action tone
no caller-id enable
!
!
! FXO
voice-port 0/3
connection plar 192
no caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern 002#T
port 0/0
huntstop
!
dial-peer voice 2 pots
destination-pattern 003#T
port 0/1
huntstop
!
dial-peer voice 3 pots
destination-pattern 3#T
port 0/2
huntstop
!
dial-peer voice 4 pots
destination-pattern 4#T
port 0/3
huntstop
!
!
!
! Voip peer configuration.
!
dial-peer voice 189 voip
destination-pattern 189
session target 10.100.0.39
session protocol sip
voice-class codec 100
vad
dtmf-relay dual-mode
!
dial-peer voice 190 voip
destination-pattern 190
session target 10.100.0.39
vad
dtmf-relay dual-mode
!
dial-peer voice 191 voip
destination-pattern 191
session target 10.100.0.39
voice-class codec 100
vad
dtmf-relay h245-alphanumeric
!
dial-peer voice 192 voip
destination-pattern 192
session target 10.100.0.39
voice-class codec 100
vad
dtmf-relay h245-alphanumeric
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.100.0.37
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 100
codec preference 1 g729
codec preference 2 g7231r63
codec preference 3 g7231r53
!
!
!
! SIP UA configuration.
!
sip-ua
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones

Так же хотелось бы увидеть конфиг AP1100F.
Проблема то собственно в нём, а не в AP1005.

AP1100F# sh run
!
version 8.30U
!
hostname AP1100F
!
!
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
!
!
no ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 10.100.0.39 255.255.255.0
!
interface ether1.0
no ip address
!
snmp name AP1100F-FXS8
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
route 0.0.0.0 0.0.0.0 10.100.0.1
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 2
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
translate-incoming called-number 100
caller-id enable
!
!
! FXS
voice-port 0/1
translate-incoming called-number 101
caller-id enable
!
!
! FXS
voice-port 0/2
translate-incoming called-number 102
caller-id enable
!
!
! FXS
voice-port 0/3
translate-incoming called-number 103
caller-id enable
!
!
! FXS
voice-port 1/0
translate-incoming called-number 104
caller-id enable
!
!
! FXS
voice-port 1/1
caller-id enable
!
!
! FXS
voice-port 1/2
caller-id enable
!
!
! FXS
voice-port 1/3
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern 189
port 0/0
huntstop
!
dial-peer voice 2 pots
destination-pattern 190
port 0/1
huntstop
!
dial-peer voice 3 pots
destination-pattern 192
port 0/2
huntstop
!
dial-peer voice 4 pots
destination-pattern 193
port 0/3
huntstop
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
destination-pattern T
session target 10.100.0.37
session protocol sip
voice-class codec 100
no vad
dtmf-relay dual-mode
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.100.0.39
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 100
codec preference 1 g729
codec preference 2 g711alaw
codec preference 3 g711ulaw
!
!
!
! Translation Rule configuration.
!
translation-rule 100
rule 0 . 002#.
!
translation-rule 101
rule 1 . 003#.
!
translation-rule 102
rule 1 . 004#.
!
translation-rule 103
rule 1 . 005#.
!
!
!
! SIP UA configuration.
!
sip-ua
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad

Вот, все так же сбрасыват при нажатии флеша, поменяли несколько разных аддпаков с разными прошивками результат один и тот же
Кстати также не ходят факсы, это тоже важно .........

добавьте в FXS шлюз
conf t
dial-peer translate-digit # !

Там косяк в новой прошивке, как исправят я выложу!

а с факсами что? bypass или t38?