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DVG-2032S проблема передачи голоса

Модераторы: Victor Kolosov, Sergei Asmankin, Sergik

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Donner

Заголовок сообщения: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 07:45

Зарегистрирован: Пн дек 20, 2010 19:37
Сообщений: 40

Схема такая 2032 - астер - нат - сиппровайдер

в схеме так же присутствую софтфоны, ip телефоны, DVG-5008S и с ними все нормально, а вот 2032 выдаёт вот что

при звонке на внешнюю линию сиппровайдеру мы слышим всё отлично, а вот нас не слышат ни в какую..... где копать?

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Alexey Motkov

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 10:46

Сотрудник D-LINK

Зарегистрирован: Пн янв 11, 2010 09:40
Сообщений: 4400

Сначала в логе.

_________________
логи можно снять
[url]http://ftp.dlink.ru/pub/VoIP/DVG-6004S/Firmware/slmon(1225112422).rar[/url]

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Donner

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 12:04

Зарегистрирован: Пн дек 20, 2010 19:37
Сообщений: 40

логи чего показать? slmon, asterisk?

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Alexey Motkov

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 12:38

Сотрудник D-LINK

Зарегистрирован: Пн янв 11, 2010 09:40
Сообщений: 4400

Лучше и то и другое.

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Donner

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 12:47

Зарегистрирован: Пн дек 20, 2010 19:37
Сообщений: 40

Код:

42:51 to connect Server, IP=<192.168.0.121 >, port=5061, local port=0 13:43:00 1039032] Receive Cmd=1100, ParaLen=5, Para=[Login]
43:00 1039032] Login == Ver(1.02.30.43.9360 2009/00/25 10:02:32] Pld(242.DLink.NoPrefix.large.vlan] Drv(0.9.5.1.1920] Hw(PNP1632-32] == 0.6.7-1
43:06 [039105] 15: Fxs Pickup
43:06 [039105] 15: ==4:GetDtmf
43:11 [039150] 15: DigitMap Route [0]
43:11 [039150] 15: Line DialNum_New [647568]
43:11 [039150] 15: Call Proxy with [647568]
43:11 [039150] 15: ==7:lnviting
43:11 [039150] 15: VoIP CallOut, Invite <sip:647568@192.160.0.129;user=phone>
43:11 [039151] 15: 1142=DIALING
43:12 [039157] 15: 1142=PROCEEDING
43:15 [039106] 15: 1142=RINGBACK
43:15 [039106] 15: ==0:RingBack
43:17 [039213] 15: 1142=RINGBACK
43:17 [039213] 15: UAC STATERING ВАСК get SDP
43:17 [039213] 15: Peer=192.160.0.129:15332, PT=0, RecvOnly=0
43:17 [039213] 15: RtpApiTalk[1,1], Peer=192.160.0.129:15332, PT=0, FC=6, NewOOB=1
43:17 [039213] 15: ==14:Talking
43:20 [039243] 15: 1142=CONNECTED
43:20 [039243] 15: St_Talking/Substatus=0
43:20 [039243] 15: Peer=192.160.0.129:15332, PT=0, RecvOnly=0 13:43:20 [039243] 15: Change RTP to 192.160.0.129:15332, pt=0
43:20 [039243] 15: RtpApiTalk[1,1], Peer=192.160.0.129:15332, PT=0, FC=6, NewOOB=1
43:25 [039292] 15: 1142=DISCONNECT
43:25 [039292] 15: Release Active Dig
43:25 [039292] 15: ==15:PlayBusyTone
43:32 [039361] DSP_ch15_check=0
43:32 [039361] 15: Fxs Hangup
43:32 [039361] 15: ==3:ldle
43:32 [039361] 15: SetFax[1]=0

это лог звонка на внешнюю линию через астер, который тишина

а это лог звонка на внутренний ip телефон, тут всё ок, слышно в обе стороны, т.е. я предполагаю что дело либо в кодеках либо нате, кодек стоит строго alaw

Код:

11:1 2 [001497] 15: Fxs Pickup 
1 1:1 2 [001497] 15: ==4:GetDtmf
1 1:16 [001537] DSP_ch15_check=0
11:16 1001537] 15: Fxs Hangup 
11:16 1001537] 15: ==3:ldle
11:16 [0015371 15: SetFax[1]=0
11:20 [0015041 15: Fxs Pickup 
11:20 [0015041 15: ==4:GetDtmf
11:24 [0016241 15: DigitMap Route [0] 
11:24 [0016241 15: Line DialNum New [666)
11:24 [0016241 15: Call Proxy with [666]
11:24 [0016241 15: ==7:lnviting
11:24 [001624) 15: VoIP CallOut, Invite <sip:666@192.160.0.129;user=phone>
11:24 [001625] 15:1142=DIALING
11:25 [001632] 15:1142=PROCEEDING
11:25 [001633] 15:1142=RINGBACK
11:25 [001633] 15: ==0:RingBack
11:20 [001657] 15:1142=CONNECTED
11:20 [001657] 15: Peer=192.160.0.129:12526, PT=0, RecvOnly=0
11:20 [001657] 15: RtpApiTalk[1,1], Peer=192.160.0.129:12526, PT=0, FC=6, NewOOB=1
11:20 [001657] 15: ==14:Talking
11:35 [001734] 15: 1142=DISCONNECT
11:35 [001734] 15: Release Active Dig
11:35 [001734] 15: ==15:PlayBusyTone

логи с астера выложу чуть попозже ибо их надо в читабельный вид приводить т.к. астер активно пользуется

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Donner

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 13:11

Зарегистрирован: Пн дек 20, 2010 19:37
Сообщений: 40

а вот лог из астериска

Код:

<--- SIP read from UDP:192.168.0.121:5060 --->
ACK sip:647568@192.168.0.129;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bKdcde4d65a4d0d67d
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>;tag=as5bd9cdec
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq:41 ACK
Max-Forwards:70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.0.121:5060 --->
INVITE sip:647568@192.168.0.129;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK9f45aa6775e5f21e
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq:42 INVITE
Contact: <sip:1142@192.168.0.121:5060>
Expires:90
Max-Forwards:70
Authorization:Digest username="1142",realm="asterisk",nonce="476ec944",uri="sip:647568@192.168.0.129;user=phone",response="9f76502760166f8cf5b19cc75e8ff3b0",algorithm=MD5
Supported: replaces
User-Agent:dlink 12-38-41929274-0.9.5.1.1928-PNP16
Content-Type: application/sdp
Content-Length: 208

v=0
o=1142 1794722930 1794722930 IN IP4 192.168.0.121
s=Session SDP
c=IN IP4 192.168.0.121
t=0 0
m=audio 9030 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16

<------------->
--- (15 headers 9 lines) ---
Sending to 192.168.0.121 : 5060 (no NAT)
Using INVITE request as basis request - D1B9-118A-466875283CA617E1D770-058@SipHost
Found peer '1142' for '1142' from 192.168.0.121:5060
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.121:9030
Looking for 647568 in phones_far (domain 192.168.0.129)
list_route: hop: <sip:1142@192.168.0.121:5060>

<--- Transmitting (no NAT) to 192.168.0.121:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK9f45aa6775e5f21e;received=192.168.0.121
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq: 42 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:647568@192.168.0.129>
Content-Length: 0


<------------>
    -- Executing [647568@phones_far:1] Dial("SIP/1142-000003dc", "SIP/neofon/647568") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 192.168.0.129 port 10536
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 62.148.237.152:5060:
INVITE sip:647568@srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.0.129:5060;branch=z9hG4bK7b8b5c83;rport
Max-Forwards: 70
From: "CIT" <sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>
Contact: <sip:206940@192.168.0.129>
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Date: Fri, 21 Jan 2011 11:04:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1888075353 1888075353 IN IP4 192.168.0.129
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.0.129
t=0 0
m=audio 10536 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called neofon/647568

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: "CIT"<sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.129:5060;received=217.8.91.6;rport=63404;branch=z9hG4bK7b8b5c83
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 407 Proxy Authentication Required
From: "CIT"<sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>;tag=808113248
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.129:5060;received=217.8.91.6;rport=63404;branch=z9hG4bK7b8b5c83
contact: <sip:647568@srgngn.usi.ru:5060;maddr=62.148.237.152>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
proxy-authenticate: Digest realm="Realm",nonce="MTI5NTYwMzUzNDA0MmI1ZTYzZDA4OTY5ZjRkNmVmMGQ4NDJjZmRkMmFlZTVi",stale=false,algorithm=MD5,qop="auth,auth-

int"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 62.148.237.152:5060:
ACK sip:647568@srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.0.129:5060;branch=z9hG4bK7b8b5c83;rport
Max-Forwards: 70
From: "CIT" <sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>;tag=808113248
Contact: <sip:206940@192.168.0.129>
all-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
Audio is at 192.168.0.129 port 10536
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 62.148.237.152:5060:
INVITE sip:647568@srgngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.0.129:5060;branch=z9hG4bK2f9756f7;rport
Max-Forwards: 70
From: "CIT" <sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>
Contact: <sip:206940@192.168.0.129>
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.2.13
Proxy-Authorization: Digest username="206940", realm="Realm", algorithm=MD5, uri="sip:647568@srgngn.usi.ru", 

nonce="MTI5NTYwMzUzNDA0MmI1ZTYzZDA4OTY5ZjRkNmVmMGQ4NDJjZmRkMmFlZTVi", response="0b0cd7a87f9e2f8d0e028490f8a79cf1", qop=auth, cnonce="007609f8", 

nc=00000001
Date: Fri, 21 Jan 2011 11:04:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1888075353 1888075354 IN IP4 192.168.0.129
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.0.129
t=0 0
m=audio 10536 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: "CIT"<sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.129:5060;received=217.8.91.6;rport=63404;branch=z9hG4bK2f9756f7
Content-Length: 0

---

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2a9c85c05155824c3bc9bc884a24e0e7@192.168.0.129' Method: OPTIONS

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 180 Ringing
From: "206940 206940"<sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>;tag=258609646
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.129:5060;received=217.8.91.6;rport=63404;branch=z9hG4bK2f9756f7
contact: <sip:647568@srgngn.usi.ru:5060;maddr=62.148.237.152>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
    -- SIP/neofon-000003dd is ringing

<--- Transmitting (no NAT) to 192.168.0.121:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK9f45aa6775e5f21e;received=192.168.0.121
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>;tag=as55c2fa1c
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq: 42 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:647568@192.168.0.129>
Content-Length: 0

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 183 Session Progress
From: "206940 206940"<sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>;tag=2276256604322011121145243
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.129:5060;received=217.8.91.6;rport=63404;branch=z9hG4bK2f9756f7
content-type: application/sdp
contact: <sip:62.148.237.152:5060;nt_end_pt=YM0+~KC0.1zQao37COn~AAf_cod~Pc9t7Vp_-Wq0bp-a.o1eGIkc17cc73cww.4oX9b~RT2Pf7z4c89Mlu15q;nt_server_host=62.148.237.152>
x-nt-party-id: -/
allow: ACK
allow: BYE
allow: CANCEL
allow: INVITE
allow: OPTIONS
allow: INFO
allow: SUBSCRIBE
allow: REFER
allow: NOTIFY
allow: PRACK
server:  CS2000_NGSS/9.0
Content-Length: 315

v=0
o=PVG 1295603497560 1295603497560 IN IP4 62.148.237.195
s=-
p=+1 6135555555
t=0 0
a=sqn: 0
a=cdsc: 1 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
m=audio 50454 RTP/AVP 8 0 101
c=IN IP4 62.148.237.195
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
--- (21 headers 14 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 62.148.237.195:50454
    -- SIP/neofon-000003dd is making progress passing it to SIP/1142-000003dc
Audio is at 192.168.0.129 port 18836
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.0.121:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK9f45aa6775e5f21e;received=192.168.0.121
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>;tag=as55c2fa1c
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq: 42 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:647568@192.168.0.129>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 22105745 22105745 IN IP4 192.168.0.129
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.0.129
t=0 0
m=audio 18836 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 200 OK
From: "206940 206940"<sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>;tag=2276256604322011121145243
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.129:5060;received=217.8.91.6;rport=63404;branch=z9hG4bK2f9756f7
content-type: application/sdp
contact: <sip:62.148.237.152:5060;nt_end_pt=YM0+~KC0.1zQao37COn~AAf_cod~Pc9t7Vp_-Wq0bp-a.o1eGIkc17cc73cww.4oX9b~RT2Pf7z4c89Mlu15q;nt_server_host=62.148.237.152>
x-nt-party-id: -/
allow: ACK,REFER
allow: BYE
allow: CANCEL
allow: INVITE
allow: OPTIONS
allow: INFO
allow: SUBSCRIBE
allow: REFER
allow: NOTIFY
allow: PRACK
x-nt-location: 193624
server:  CS2000_NGSS/9.0
Content-Length: 315

v=0
o=PVG 1295603497560 1295603497560 IN IP4 62.148.237.195
s=-
p=+1 6135555555
t=0 0
a=sqn: 0
a=cdsc: 1 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
m=audio 50454 RTP/AVP 8 0 101
c=IN IP4 62.148.237.195
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
--- (22 headers 14 lines) ---
list_route: hop: <sip:62.148.237.152:5060;nt_end_pt=YM0+~KC0.1zQao37COn~AAf_cod~Pc9t7Vp_-Wq0bp-

a.o1eGIkc17cc73cww.4oX9b~RT2Pf7z4c89Mlu15q;nt_server_host=62.148.237.152>
set_destination: Parsing <sip:62.148.237.152:5060;nt_end_pt=YM0+~KC0.1zQao37COn~AAf_cod~Pc9t7Vp_-Wq0bp-

a.o1eGIkc17cc73cww.4oX9b~RT2Pf7z4c89Mlu15q;nt_server_host=62.148.237.152> for address/port to send to
set_destination: set destination to 62.148.237.152, port 5060
Transmitting (NAT) to 62.148.237.152:5060:
ACK sip:62.148.237.152:5060;nt_end_pt=YM0+~KC0.1zQao37COn~AAf_cod~Pc9t7Vp_-Wq0bp-a.o1eGIkc17cc73cww.4oX9b~RT2Pf7z4c89Mlu15q;nt_server_host=62.148.237.152 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.129:5060;branch=z9hG4bK26dfd382;rport
Max-Forwards: 70
From: "CIT" <sip:206940@srgngn.usi.ru>;tag=as1162788e
To: <sip:647568@srgngn.usi.ru>;tag=2276256604322011121145243
Contact: <sip:206940@192.168.0.129>
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.2.13
Content-Length: 0


---
    -- SIP/neofon-000003dd answered SIP/1142-000003dc
Audio is at 192.168.0.129 port 18836
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.121:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK9f45aa6775e5f21e;received=192.168.0.121
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>;tag=as55c2fa1c
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq: 42 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:647568@192.168.0.129>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 22105745 22105746 IN IP4 192.168.0.129
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.0.129
t=0 0
m=audio 18836 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/1142-000003dc and SIP/neofon-000003dd

<--- SIP read from UDP:192.168.0.121:5060 --->
ACK sip:647568@192.168.0.129 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.121:5060;branch=z9hG4bK7dff1f49975ab4b7
From: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
To: <sip:647568@192.168.0.129;user=phone>;tag=as55c2fa1c
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq:42 ACK
Max-Forwards:70
Authorization:Digest username="1142",realm="asterisk",nonce="476ec944",uri="sip:647568@192.168.0.129;user=phone",response="9f76502760166f8cf5b19cc75e8ff3b0",algorithm=MD5
User-Agent:dlink 12-38-41929274-0.9.5.1.1928-PNP16
Content-Length: 0


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

<--- SIP read from UDP:62.148.237.152:5060 --->
BYE sip:206940@192.168.0.129:63404;maddr=217.8.91.6 SIP/2.0
From: <sip:647568@srgngn.usi.ru>;tag=2276256604322011121145243
To: "206940 206940"<sip:206940@srgngn.usi.ru>;tag=as1162788e
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 104 BYE
Via: SIP/2.0/UDP 62.148.237.152:5060;branch=z9hG4bK-4289c0-3ea1793-57af204d
user-agent:  CS2000_NGSS/9.0
max-forwards: 69
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
allow: ACK
allow: BYE
allow: CANCEL
allow: INVITE
allow: OPTIONS
allow: INFO
allow: SUBSCRIBE
allow: REFER
allow: NOTIFY
allow: PRACK
allow: UPDATE
Content-Length: 0


<------------->
--- (21 headers 0 lines) ---
Sending to 62.148.237.152 : 5060 (NAT)

<--- Transmitting (NAT) to 62.148.237.152:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.148.237.152:5060;branch=z9hG4bK-4289c0-3ea1793-57af204d;received=62.148.237.152
From: <sip:647568@srgngn.usi.ru>;tag=2276256604322011121145243
To: "206940 206940"<sip:206940@srgngn.usi.ru>;tag=as1162788e
Call-ID: 2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru
CSeq: 104 BYE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (phones_far, 647568, 1) exited non-zero on 'SIP/1142-000003dc'
Scheduling destruction of SIP dialog 'D1B9-118A-466875283CA617E1D770-058@SipHost' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:1142@192.168.0.121:5060> for address/port to send to
set_destination: set destination to 192.168.0.121, port 5060
Reliably Transmitting (no NAT) to 192.168.0.121:5060:
BYE sip:1142@192.168.0.121:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.129:5060;branch=z9hG4bK2609d6a1;rport
Max-Forwards: 70
From: <sip:647568@192.168.0.129;user=phone>;tag=as55c2fa1c
To: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.121:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.129:5060;rport;branch=z9hG4bK2609d6a1
From: <sip:647568@192.168.0.129;user=phone>;tag=as55c2fa1c
To: "1142" <sip:1142@192.168.0.129>;tag=51f6998a-687528
Call-ID: D1B9-118A-466875283CA617E1D770-058@SipHost
CSeq:102 BYE
User-Agent:dlink 12-38-41929274-0.9.5.1.1928-PNP16
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'D1B9-118A-466875283CA617E1D770-058@SipHost' Method: ACK
Really destroying SIP dialog '2f29d39a063753ea10524ca91a5d9780@srgngn.usi.ru' Method: BYE

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Donner

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Пт янв 21, 2011 14:01

Зарегистрирован: Пн дек 20, 2010 19:37
Сообщений: 40

попробовал переключить на кодек g726 - заработало, но качество куда хуже.... т.е. это не решение... может есть прошивка где кодек нормально обрабатывается?

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Donner

Заголовок сообщения: Re: DVG-2032S проблема передачи голоса

Добавлено: Сб янв 22, 2011 09:27

Зарегистрирован: Пн дек 20, 2010 19:37
Сообщений: 40

вроде бы решил проблему.... порыл сайт и нашел вариант что нужно поменять значение dtmf в sip_info на шлюзе, поменял и всё заработало, как это связано с самой проблемой остаётся загадкой.

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