Ребята, подскажите, есть ли решение подобной проблемы:
настроен шлюз d-link DVG-7062S (http://dlink.ru/ru/faq/95/547.html), Asteriks, 6 FXS + 2 FXO всё работает нормально, но когда совершаешь исходящий звонок, то переодически возникает ситуация, когда на том конце слышат голос, а у нас в трубке тишина. (в момент времени, когда обычно начинаются длинные гудки в трубке, в случае неудачи: ничего не слышно, то есть полностью отрубается звук, но вызов на номер идет, и другие абоненты его принимают и могут нас слышать, а мы не слышим ни гудка, ни голоса абонента, когда он подымает трубку). Перепрошивка dlink DVG-7062S данную проблему не решила.
Спасибо.

Выложите пожалуйста лог

есть только лог asterisk:
первый звонок прошел нормально, а второй с симптомами того, что мы ничего не слышали в трубке, а нас слышали:
,"01201","89242830732","officeSU","""New User"" <01201>","SIP/01201-08286dc0","SIP/TRUNK_DLINK-082b0b00","Dial","SIP/4242464654@TRUNK_DLINK,,tTmM(send-num,89242830732)","2010-04-26 06:49:33","2010-04-26 06:49:33","2010-04-26 06:49:52",19,19,"ANSWERED","DOCUMENTATION","1272264573.593",""
,"01201","89242830732","officeSU","""New User"" <01201>","SIP/01201-082b0b00","SIP/TRUNK_DLINK-08286dc0","Dial","SIP/4242464654@TRUNK_DLINK,,tTmM(send-num,89242830732)","2010-04-26 06:50:03","2010-04-26 06:50:03","2010-04-26 06:50:28",25,25,"ANSWERED","DOCUMENTATION","1272264603.595",""

Выложите лог шлюза.
Лог снимается SLmon'ом:
ftp://dlink.ru/pub/VoIP/DVG-6004S/Firmw ... 1225112422).rar

спасибо за прогу!
Первый звонок, в 10:37:47 прошел с проблемой (мы не слышали звук),
второй, в 10:38:32 прошел нормально (со слышимым звуком с обоих сторон)

10:36:32 [756310] Receive Cmd=1100, ParaLen=5, Para=[Login]
10:36:32 [756310] Login == Ver(1.02.38.43.9367 2009/08/25 17:45:23) PId(242.DLink.NoPrefix.drtp.vlan) Drv(0.9.5.1.1928) Hw(PBX6S2O) == 0.6.7-1
10:36:40 [756387] 0: Fxs Pickup
10:37:47 [757062] 0: ==4:GetDtmf
10:37:53 [757118] 0: End Input Default Route [0]
10:37:53 [757118] 0: Line DialNum_New [89242830732]
10:37:53 [757118] 0: Call Proxy with [89242830732]
10:37:53 [757119] 0: ==7:Inviting
10:37:53 [757119] 0: VoIP CallOut, Invite <sip:89242830732@192.168.121.2;user=phone>
10:37:53 [757120] 0: 01201=DIALING
10:37:53 [757122] 0: 01201=PROCEEDING
10:37:53 [757123] 6: 4242464654=OFFERING
10:37:53 [757123] 6: Get CallerId=New User
10:37:53 [757123] 6: Peer=192.168.121.2:10388, PT=8, RecvOnly=0
10:37:53 [757123] 6: TrunkPrefix=, Dest=, Dialno=
10:37:54 [757123] 6: FxoHookOff
10:37:54 [757123] 6: SetInputGain(4)
10:37:54 [757123] 6: ==18:TrunkDialOut
10:37:54 [757124] 0: 01201=RINGBACK
10:37:54 [757124] 0: UACSTATE_RINGBACK get SDP
10:37:54 [757124] 0: Peer=192.168.121.2:10898, PT=8, RecvOnly=0
10:37:54 [757124] 0: RtpApiTalk[1,1], Peer=192.168.121.2:10898, PT=8, FC=2, NewOOB=1
10:37:54 [757124] 0: ==14:Talking
10:37:54 [757124] 6: Fxo DialOut OK
10:37:54 [757125] 6: 4242464654=ACCEPT
10:37:54 [757125] 6: ==13:VoipAnswering
10:37:54 [757126] 6: 4242464654=CONNECTED
10:37:54 [757126] 6: RtpApiTalk[1,1], Peer=192.168.121.2:10388, PT=8, FC=2, NewOOB=1
10:37:54 [757126] 6: ==14:Talking
10:37:57 [757154] 6: Get(2)
10:37:57 [757157] 6: Get(4)
10:37:59 [757182] 0: 01201=CONNECTED
10:37:59 [757182] 0: St_Talking/Substatus=0
10:37:59 [757182] 0: Peer=192.168.121.2:10898, PT=8, RecvOnly=0
10:37:59 [757182] 0: Change RTP to 192.168.121.2:10898, pt=8
10:37:59 [757182] 0: RtpApiTalk[1,1], Peer=192.168.121.2:10898, PT=8, FC=2, NewOOB=1
10:38:26 [757445] 6: Fxo Detected CPT Busy(10)
10:38:26 [757445] DSP_ch6_check=0
10:38:26 [757445] 6: FxoHookOn
10:38:26 [757445] 6: Bye at Talking
10:38:26 [757445] 6: ==15:PlayBusyTone
10:38:26 [757446] DSP_ch6_check=0
10:38:26 [757446] 6: FxoHookOn
10:38:26 [757446] 6: ==3:Idle
10:38:26 [757446] 6: SetInputGain(4)
10:38:26 [757446] 6: SetFax(1)=0
10:38:26 [757446] 6: 4242464654=DISCONNECT
10:38:26 [757446] 6: Release Old Dlg #2
10:38:26 [757447] 0: 01201=DISCONNECT
10:38:26 [757447] 0: Release Active Dlg
10:38:26 [757447] 0: ==15:PlayBusyTone
10:38:29 [757476] DSP_ch0_check=0
10:38:29 [757476] 0: Fxs Hangup
10:38:29 [757476] 0: ==3:Idle
10:38:29 [757476] 0: SetFax(1)=0
10:38:32 [757507] 0: Fxs Pickup
10:38:32 [757507] 0: ==4:GetDtmf
10:38:36 [757550] 0: End Input Default Route [0]
10:38:36 [757550] 0: Line DialNum_New [89242830732]
10:38:36 [757550] 0: Call Proxy with [89242830732]
10:38:36 [757550] 0: ==7:Inviting
10:38:36 [757550] 0: VoIP CallOut, Invite <sip:89242830732@192.168.121.2;user=phone>
10:38:36 [757551] 0: 01201=DIALING
10:38:36 [757553] 0: 01201=PROCEEDING
10:38:37 [757554] 6: 4242464654=OFFERING
10:38:37 [757554] 6: Get CallerId=New User
10:38:37 [757554] 6: Peer=192.168.121.2:11366, PT=8, RecvOnly=0
10:38:37 [757554] 6: TrunkPrefix=, Dest=, Dialno=
10:38:37 [757554] 6: FxoHookOff
10:38:37 [757554] 6: SetInputGain(4)
10:38:37 [757554] 6: ==18:TrunkDialOut
10:38:37 [757555] 0: 01201=RINGBACK
10:38:37 [757555] 0: UACSTATE_RINGBACK get SDP
10:38:37 [757555] 0: Peer=192.168.121.2:12388, PT=8, RecvOnly=0
10:38:37 [757555] 0: RtpApiTalk[1,1], Peer=192.168.121.2:12388, PT=8, FC=2, NewOOB=1
10:38:37 [757555] 0: ==14:Talking
10:38:37 [757555] 6: Fxo DialOut OK
10:38:37 [757556] 6: 4242464654=ACCEPT
10:38:37 [757556] 6: ==13:VoipAnswering
10:38:37 [757557] 6: 4242464654=CONNECTED
10:38:37 [757557] 6: RtpApiTalk[1,1], Peer=192.168.121.2:11366, PT=8, FC=2, NewOOB=1
10:38:37 [757557] 6: ==14:Talking
10:38:42 [757613] 0: 01201=CONNECTED
10:38:43 [757613] 0: St_Talking/Substatus=0
10:38:43 [757613] 0: Peer=192.168.121.2:12388, PT=8, RecvOnly=0
10:38:43 [757613] 0: Change RTP to 192.168.121.2:12388, pt=8
10:38:43 [757613] 0: RtpApiTalk[1,1], Peer=192.168.121.2:12388, PT=8, FC=2, NewOOB=1
10:39:09 [757877] DSP_ch0_check=0
10:39:09 [757877] 0: Fxs Hangup
10:39:09 [757877] 0: Bye at Talking
10:39:09 [757877] 0: ==3:Idle
10:39:09 [757878] 0: SetFax(1)=0
10:39:09 [757878] 0: 01201=DISCONNECT
10:39:09 [757878] 0: Release Old Dlg #2
10:39:09 [757879] 6: 4242464654=DISCONNECT
10:39:09 [757879] 6: Release Active Dlg
10:39:09 [757879] 6: ==15:PlayBusyTone
10:39:09 [757880] 6: Start Btc Learning
10:39:12 [757904] 6: Fxo Detected CPT Busy(10)
10:39:12 [757904] DSP_ch6_check=0
10:39:12 [757904] 6: FxoHookOn
10:39:12 [757904] DSP_ch6_check=0
10:39:12 [757904] 6: FxoHookOn
10:39:12 [757904] 6: ==3:Idle
10:39:12 [757904] 6: SetInputGain(4)
10:39:12 [757904] 6: SetFax(1)=0

если в логе появляется get(2) или get(4) или get(9), то звук пропадает. как в первом звонке 10:37:57 [757154] 6: Get(2)

Вот еще лог, который ведет сам шлюз:

Это нормальный звонок:

Apr 27 11:12:46 192.168.121.4 001E5842EE7C-sipg: New User CallIn
Apr 27 11:12:46 192.168.121.4 001E5842EE7C-sipg: 6: TrunkDialOut
Apr 27 11:12:46 192.168.121.4 001E5842EE7C-sipg: 0: Talking
Apr 27 11:12:46 192.168.121.4 001E5842EE7C-sipg: 6: PSTN Dial
Apr 27 11:12:47 192.168.121.4 001E5842EE7C-sipg: 6: VoipAnswering
Apr 27 11:12:47 192.168.121.4 001E5842EE7C-sipg: 6: Connected As Callee
Apr 27 11:12:47 192.168.121.4 001E5842EE7C-sipg: 6: Talking
Apr 27 11:12:52 192.168.121.4 001E5842EE7C-sipg: 0: Connected As Caller
Apr 27 11:13:19 192.168.121.4 001E5842EE7C-sipg: 0: RTP Lost 19/100 (297 pkt)
Apr 27 11:13:20 192.168.121.4 001E5842EE7C-sipg: 0: Released By Me
Apr 27 11:13:20 192.168.121.4 001E5842EE7C-sipg: 0: Idle
Apr 27 11:13:20 192.168.121.4 001E5842EE7C-sipg: 6: Released By Peer
Apr 27 11:13:20 192.168.121.4 001E5842EE7C-sipg: 6: PlayBusyTone
Apr 27 11:13:20 192.168.121.4 001E5842EE7C-sipg: 6: Idle

А вот это не нормальный:

Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: New User CallIn
Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: 6: TrunkDialOut
Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: 0: Talking
Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: 6: PSTN Dial
Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: 6: VoipAnswering
Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: 6: Connected As Callee
Apr 27 11:14:33 192.168.121.4 001E5842EE7C-sipg: 6: Talking
Apr 27 11:14:39 192.168.121.4 001E5842EE7C-sipg: 0: Connected As Caller
Apr 27 11:15:01 192.168.121.4 001E5842EE7C-sipg: 0: Released By Me
Apr 27 11:15:02 192.168.121.4 001E5842EE7C-sipg: 0: Idle
Apr 27 11:15:02 192.168.121.4 001E5842EE7C-sipg: 6: Released By Peer
Apr 27 11:15:02 192.168.121.4 001E5842EE7C-sipg: 6: PlayBusyTone
Apr 27 11:15:04 192.168.121.4 001E5842EE7C-sipg: 6: Idle

Если связь с get() (FXO услышал тоны клавиш) прослеживается, то можно предположить высокую чувствительность DTMF. Уменьшите ее.

уменьшил DTMF Detection Sensitivity, но проблема осталась. Как выяснил в процессе, иногда и при присутствии get() в логе, вызов срабатывает нормально, но в основном нет. Если get отсутствует, то вызов точно проходит хорошо.

Больше в логах ничего криминального я не нашел.
Давайте предположим непрохождение ртп трафика. Какая конфигурация сети? Файрволы? Шлюзы?
И давайте лог Астера с включенным sip debug.

Вот лог проблемного звонка:

<------------->
--- (10 headers 0 lines) ---
oes-sakh-nestle*CLI>
<--- SIP read from UDP://192.168.121.4:5060 --->
BYE sip:89242830732@192.168.121.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bKfa9bf85dc4019d03
From: "01201" <sip:01201@192.168.121.2>;tag=6c8b8896-687303
To: <sip:89242830732@192.168.121.2;user=phone>;tag=as53f329ff
Call-ID: D1B9-8A15-46687303FA18C09ECE27-045@SipHost
CSeq:36 BYE
Max-Forwards:70
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.121.4 : 5060 (no NAT)
oes-sakh-nestle*CLI>
<--- Transmitting (no NAT) to 192.168.121.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bKfa9bf85dc4019d03;received=192.168.121.4
From: "01201" <sip:01201@192.168.121.2>;tag=6c8b8896-687303
To: <sip:89242830732@192.168.121.2;user=phone>;tag=as53f329ff
Call-ID: D1B9-8A15-46687303FA18C09ECE27-045@SipHost
CSeq: 36 BYE
Server: Asterisk PBX 1.6.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '3eb0c4e00e9e7dc85bdcc2a371450bb7@192.168.121.2' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:4242464654@192.168.121.4:5060> for address/port to send to
set_destination: set destination to 192.168.121.4, port 5060
Reliably Transmitting (no NAT) to 192.168.121.4:5060:
BYE sip:4242464654@192.168.121.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.2:5060;branch=z9hG4bK645cd6cc
Max-Forwards: 70
From: "New User" <sip:01201@192.168.121.2>;tag=as49a91b04
To: <sip:4242464654@192.168.121.4>;tag=b9e27619-687304
Call-ID: 3eb0c4e00e9e7dc85bdcc2a371450bb7@192.168.121.2
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.1.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
oes-sakh-nestle*CLI>
<--- SIP read from UDP://192.168.121.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.121.2:5060;branch=z9hG4bK645cd6cc
From: "New User" <sip:01201@192.168.121.2>;tag=as49a91b04
To: <sip:4242464654@192.168.121.4>;tag=b9e27619-687304
Call-ID: 3eb0c4e00e9e7dc85bdcc2a371450bb7@192.168.121.2
CSeq:103 BYE
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3eb0c4e00e9e7dc85bdcc2a371450bb7@192.168.121.2' Method: INVITE
Really destroying SIP dialog 'D1B9-8A15-46687303FA18C09ECE27-045@SipHost' Method: BYE
oes-sakh-nestle*CLI>
<--- SIP read from UDP://192.168.121.4:5060 --->
BYE sip:01201@192.168.121.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bKa71e3a7a5ea5ba8f
From: <sip:4242464654@192.168.121.4>;tag=b9e27619-687304
To: "New User" <sip:01201@192.168.121.2>;tag=as49a91b04
Call-ID: 3eb0c4e00e9e7dc85bdcc2a371450bb7@192.168.121.2
CSeq:9 BYE
Contact: <sip:4242464654@192.168.121.4:5060>
Max-Forwards:70
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
oes-sakh-nestle*CLI>
<--- Transmitting (no NAT) to 192.168.121.4:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bKa71e3a7a5ea5ba8f;received=192.168.121.4
From: <sip:4242464654@192.168.121.4>;tag=b9e27619-687304
To: "New User" <sip:01201@192.168.121.2>;tag=as49a91b04
Call-ID: 3eb0c4e00e9e7dc85bdcc2a371450bb7@192.168.121.2
CSeq: 9 BYE
Server: Asterisk PBX 1.6.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>




А это лог нормального звонка:

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1d2fe2f719a4d0040bd6748a3db9dcc8@192.168.121.2' Method: INVITE
Really destroying SIP dialog 'D1B9-8A15-46687869515CE84D2C26-050@SipHost' Method: BYE
Audio is at 192.168.121.2 port 14968
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.121.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bK291c12fe89575c27;received=192.168.121.4
From: "01201" <sip:01201@192.168.121.2>;tag=9cf757ee-687919
To: <sip:89242830732@192.168.121.2;user=phone>;tag=as249d956e
Call-ID: D1B9-8A15-466879199580B99EDD3F-051@SipHost
CSeq: 40 INVITE
Server: Asterisk PBX 1.6.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:89242830732@192.168.121.2>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1902282787 1902282788 IN IP4 192.168.121.2
s=Asterisk PBX 1.6.1.2
c=IN IP4 192.168.121.2
t=0 0
m=audio 14968 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
oes-sakh-nestle*CLI>
<--- SIP read from UDP://192.168.121.4:5060 --->
ACK sip:89242830732@192.168.121.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bK19e5e882de3e05ab
From: "01201" <sip:01201@192.168.121.2>;tag=9cf757ee-687919
To: <sip:89242830732@192.168.121.2;user=phone>;tag=as249d956e
Call-ID: D1B9-8A15-466879199580B99EDD3F-051@SipHost
CSeq:40 ACK
Max-Forwards:70
Authorization:Digest username="01201",realm="asterisk",nonce="6506f21e",uri="sip:89242830732@192.168.121.2;user=phone",response="7f65d4413b6fb190d52c7d4a64648a44",algorithm=MD5
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
oes-sakh-nestle*CLI>
<--- SIP read from UDP://192.168.121.4:5060 --->
BYE sip:89242830732@192.168.121.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bKe16f6a3c99474ff5
From: "01201" <sip:01201@192.168.121.2>;tag=9cf757ee-687919
To: <sip:89242830732@192.168.121.2;user=phone>;tag=as249d956e
Call-ID: D1B9-8A15-466879199580B99EDD3F-051@SipHost
CSeq:41 BYE
Max-Forwards:70
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.121.4 : 5060 (no NAT)
oes-sakh-nestle*CLI>
<--- Transmitting (no NAT) to 192.168.121.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.121.4:5060;branch=z9hG4bKe16f6a3c99474ff5;received=192.168.121.4
From: "01201" <sip:01201@192.168.121.2>;tag=9cf757ee-687919
To: <sip:89242830732@192.168.121.2;user=phone>;tag=as249d956e
Call-ID: D1B9-8A15-466879199580B99EDD3F-051@SipHost
CSeq: 41 BYE
Server: Asterisk PBX 1.6.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '06625bed79405cf25ff5b6d74ed25ed9@192.168.121.2' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:4242464654@192.168.121.4:5060> for address/port to send to
set_destination: set destination to 192.168.121.4, port 5060
Reliably Transmitting (no NAT) to 192.168.121.4:5060:
BYE sip:4242464654@192.168.121.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.121.2:5060;branch=z9hG4bK18600e57
Max-Forwards: 70
From: "New User" <sip:01201@192.168.121.2>;tag=as76462219
To: <sip:4242464654@192.168.121.4>;tag=42a8199d-687920
Call-ID: 06625bed79405cf25ff5b6d74ed25ed9@192.168.121.2
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.1.2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
oes-sakh-nestle*CLI>
<--- SIP read from UDP://192.168.121.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.121.2:5060;branch=z9hG4bK18600e57
From: "New User" <sip:01201@192.168.121.2>;tag=as76462219
To: <sip:4242464654@192.168.121.4>;tag=42a8199d-687920
Call-ID: 06625bed79405cf25ff5b6d74ed25ed9@192.168.121.2
CSeq:103 BYE
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '06625bed79405cf25ff5b6d74ed25ed9@192.168.121.2' Method: INVITE
Really destroying SIP dialog 'D1B9-8A15-466879199580B99EDD3F-051@SipHost' Method: BYE

Это только хвосты звонков и из разных мест.
Видимо 192.168.121.2 — это астер, 192.168.121.4 — это шлюз.

да =)

подскажите, пожалуйста, где его взять?

/var/log/asterisk
как-то так...