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Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

0

Aterisk стоит на PFsense по внутренней IP телефонии(да и по внешней тоже) идет жуткое эхо если у вашего оппонента по звонку стоит Grandstream DP710/715. Причем если у оппонента трубка Panasonic, регистренная на базе того же грандстрима D715 - эха почти нет. Так же нет эха на грандстримах GXP2124 и USB телефонах SkypeMate.

sip.cfg настройка Grandstreamov

[gs715](!) ; <== обратите внимание, восклицательный знак
; взят в круглые скобки. Это признак шаблона.
type=friend
context=phones
host=dynamic
disallow=all
allow=ulaw
dtmfmode=rfc2833
secret=qwerty

[xlite](!)
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
"Transmit Silence"=YES
type=friend
;regexten=1234                   ; When they register, create extension 1234
context=phones
host=dynamic                    ; This device needs to register
secret=qwerty
directmedia=no                  ; Typically set to NO if behind NAT
disallow=all
allow=gsm                       ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw

[101](gs715)
callerid = "Secretary" <101> 
callgroup=1
pickupgroup=1

[103](gs715)
callerid = "user103" <103> 
callgroup=1
pickupgroup=1

[201](xlite)
callerid = "User 201" <201> 
callgroup=1
pickupgroup=1

......

extension.cfg

[general]
autofallthrough=yes

[phones]
include => outbound-local
include => outbound-global
include => employees

[employees]
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,HangUp()
exten => _2XX,1,Dial(SIP/${EXTEN})
exten => _2XX,n,HangUp()

........

в настроках базы использую по приоритету кодеки: PCMU/PCMA

не пойму, где копать и кто тут виноват....трубки, базы или сам астериск...

ЛОГИ по пиру 110 (как раз GP710)

<--- SIP read from UDP:192.168.1.204:5062 --->
INVITE sip:310@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK74251880;rport
Route: <sip:192.168.1.100:5060;lr>
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 150 INVITE
Contact: "User 110" <sip:110@192.168.1.204:5062>
Max-Forwards: 70
User-Agent: Grandstream DP715 1.0.0.23
Privacy: none
P-Preferred-Identity: "User 110" <sip:110@192.168.1.100>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 388

v=0
o=110 8002 8000 IN IP4 192.168.1.204
s=SIP Call
c=IN IP4 192.168.1.204
t=0 0
m=audio 5008 RTP/AVP 0 8 4 18 2 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=r--- (17 headers 18 lines) ---
Sending to 192.168.1.204:5062 (NAT)
Using INVITE request as basis request - 678941221-5062-16@BJC.BGI.B.CAE
Found peer '110' for '110' from 192.168.1.204:5062

<--- Reliably Transmitting (NAT) to 192.168.1.204:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK74251880;received=192.168.1.204;rport=5062
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>;tag=as51d966dd
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 150 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7c3f36a8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '678941221-5062-16@BJC.BGI.B.CAE' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.204:5062 --->
ACK sip:310@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK74251880;rport
Route: <sip:192.168.1.100:5060;lr>
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>;tag=as51d966dd
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 150 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.204:5062 --->
INVITE sip:310@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;rport
Route: <sip:192.168.1.100:5060;lr>
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 151 INVITE
Contact: "User 110" <sip:110@192.168.1.204:5062>
Authorization: Digest username="110", realm="asterisk", nonce="7c3f36a8", uri="sip:310@192.168.1.100", response="cbd9f3618752a9c76c4ea3d23922377a", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream DP715 1.0.0.23
Privacy: none
P-Preferred-Identity: "User 110" <sip:110@192.168.1.100>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 388

v=0
o=110 8002 8000 IN IP4 192.168.1.204
s=SIP Call
c=IN IP4 192.168.1.204
t=0 0
m=audio 5008 RTP/AVP 0 8 4 18--- (18 headers 18 lines) ---
Sending to 192.168.1.204:5062 (NAT)
Using INVITE request as basis request - 678941221-5062-16@BJC.BGI.B.CAE
Found peer '110' for '110' from 192.168.1.204:5062
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.204:5008
Looking for 310 in phones (domain 192.168.1.100)
list_route: hop: <sip:110@192.168.1.204:5062>

<--- Transmitting (NAT) to 192.168.1.204:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;received=192.168.1.204;rport=5062
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 151 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:310@192.168.1.100:5060>
Content-Length: 0


<------------>
    -- Executing [310@phones:1] Dial("SIP/110-00000168", "SIP/310") in new stack
    -- Called SIP/310
    -- SIP/310-00000169 is ringing

<--- Transmitting (NAT) to 192.168.1.204:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;received=192.168.1.204;rport=5062
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>;tag=as5369cdd3
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 151 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:310@192.168.1.100:5060>
Content-Length: 0


<------------>
    -- SIP/310-00000169 answered SIP/110-00000168
Audio is at 13062
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.204:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;received=192.168.1.204;rport=5062
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>;tag=as5369cdd3
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 151 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:310@192.168.1.100:5060>
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 1635771721 1635771721 IN IP4 192.168.1.100
s=Asterisk PBX 1.8.19.0
c=IN IP4 192.168.1.100
t=0 0
m=audio 13062 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/110-00000168 and SIP/310-00000169

<--- SIP read from UDP:192.168.1.204:5062 --->
ACK sip:310@192.168.1.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK623808183;rport
From: "User 110" <sip:110@192.168.1.100>;tag=1842811006
To: <sip:310@192.168.1.100>;tag=as5369cdd3
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 151 ACK
Contact: <sip:110@192.168.1.204:5062>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
  == Spawn extension (phones, 310, 1) exited non-zero on 'SIP/110-00000168'
Scheduling destruction of SIP dialog '678941221-5062-16@BJC.BGI.B.CAE' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:110@192.168.1.204:5062> for address/port to send to
set_destination: set destination to 192.168.1.204:5062
Reliably Transmitting (NAT) to 192.168.1.204:5062:
BYE sip:110@192.168.1.204:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK766d9793;rport
Max-Forwards: 70
From: <sip:310@192.168.1.100>;tag=as5369cdd3
To: "User 110" <sip:110@192.168.1.100>;tag=1842811006
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.19.0
Proxy-Authorization: Digest username="110", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.100", nonce="", response="e4f3d1f63862a146ddcb56ddbc147491"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.204:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK766d9793;rport=5060
From: <sip:310@192.168.1.100>;tag=as5369cdd3
To: "User 110" <sip:110@192.168.1.100>;tag=1842811006
Call-ID: 678941221-5062-16@BJC.BGI.B.CAE
CSeq: 102 BYE
Contact: <sip:110@192.168.1.204:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream DP715 1.0.0.23
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '678941221-5062-16@BJC.BGI.B.CAE' Method: ACK
удалить закрыть спам изменить тег редактировать

спросил 2013-11-21 12:52:50 +0400

Menog Gravatar Menog
1 2 1

обновил 2013-11-21 13:54:35 +0400

Comments

Путь прохождения вызова покажите.

switch ( 2013-11-21 12:55:19 +0400 )редактировать

всмысле путь? логи какие-то?

в первом сообщении подправил.

Menog ( 2013-11-21 13:33:17 +0400 )редактировать

нет, как и через что проходит вызов.

switch ( 2013-11-21 14:03:03 +0400 )редактировать

вызов через Asterisk Pfsense сервер 192,168,1,100 в логах - 110 юзер(Grandstream DP715) звонит 310-му (SkypeMate телефон) оба юзера подключены к одному комутатору который соответственно подключен к другому коммутатору в 20 метрах, в который соответственно воткнут астериск сервер...т.к. задержки должны быть минимальные

Menog ( 2013-11-21 14:15:09 +0400 )редактировать

а зачем у вас NAT в этой схеме включен ? Прошивку обновляли ?

awsswa ( 2013-11-21 15:30:38 +0400 )редактировать

Действительно NAT тут совсем не нужен, отключил для всех пиров. Астерист как и пфсенс последний стоит....

Menog ( 2013-11-21 16:25:54 +0400 )редактировать

эхо чистое, skypemate не дает эха, т.к. при тестировании впаре с теми же панасониками нигде эха нет. прошивку пробовал - не помогло.

Menog ( 2013-11-22 09:41:17 +0400 )редактировать

2 Ответа

0

Эхо чистое или с бульканьем, щелчками?

Пошивку другую пробовал?

ссылка удалить спам редактировать

ответил 2013-11-21 17:25:15 +0400

bolshoy_plohish Gravatar bolshoy_plohish
1358 21 16 37
0

у вас SkypeMate телефон и дает эхо. У него врядли есть какой-либо эходав.

ссылка удалить спам редактировать

ответил 2013-11-21 15:37:10 +0400

switch Gravatar switch
8334 11 7 91
http://lynks.ru/

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Задан: 2013-11-21 12:52:50 +0400

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Обновлен: Nov 21 '13

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Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.