История изменений - Asterisk: Вопросы и Ответы

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

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

в настроках базы использую по приоритету кодеки: PCMU/PCMA

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

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

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

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

[103](gs715)
callerid = "user103" <103> 
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...

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

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...

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

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...

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <sip:110@192.168.1.204:5062> Max-Forwards: 70 User-Agent: Grandstream DP715 1.0.0.23 Privacy: none P-Preferred-Identity: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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) setdestination: Parsing <sip:110@192.168.1.204:5062> for address/port to send to setdestination: 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: "Tolstobrov A.Y." <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: "Tolstobrov A.Y." <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

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

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 from UDP:192.168.1.204:5062 ---> INVITE ---> INVITE sip:310@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 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: "Tolstobrov A.Y." "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: <sip:310@192.168.1.100> Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 150 INVITE INVITE Contact: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.204:5062> Max-Forwards: 70 User-Agent: 70 User-Agent: Grandstream DP715 1.0.0.23 Privacy: none 1.0.0.23 Privacy: none P-Preferred-Identity: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100> Supported: replaces, path, timer, eventlist timer, eventlist Allow: INVITE, ACK, OPTIONS, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 388 v=0 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 IP4 192.168.1.204 s=SIP Call Call c=IN IP4 192.168.1.204 192.168.1.204 t=0 0 0 m=audio 5008 RTP/AVP 0 8 4 18 2 97 101 101 a=sendrecv a=rtpmap:0 PCMU/8000 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 annexb=no a=rtpmap:2 G726-32/8000 a=r--- (17 headers 18 18 lines) --- --- Sending to to 192.168.1.204:5062 (NAT) (NAT) Using INVITE request as basis request - 678941221-5062-16@BJC.BGI.B.CAE Found - 678941221-5062-16@BJC.BGI.B.CAE Found peer '110' for '110' from 192.168.1.204:5062 from 192.168.1.204:5062 <--- Reliably Transmitting (NAT) to to 192.168.1.204:5062 ---> ---> SIP/2.0 401 Unauthorized Unauthorized Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK74251880;received=192.168.1.204;rport=5062 From: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 To: To: <sip:310@192.168.1.100>;tag=as51d966dd Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 150 INVITE INVITE Server: Asterisk PBX 1.8.19.0 PBX 1.8.19.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: PUBLISH Supported: replaces, timer WWW-Authenticate: timer WWW-Authenticate: Digest algorithm=MD5, algorithm=MD5, realm="asterisk", nonce="7c3f36a8" Content-Length: 0 <------------> 0 <------------> Scheduling destruction destruction of SIP dialog dialog '678941221-5062-16@BJC.BGI.B.CAE' in in 32000 ms (Method: INVITE) INVITE) <--- SIP read from from UDP:192.168.1.204:5062 ---> ACK ---> ACK sip:310@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 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: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 To: To: <sip:310@192.168.1.100>;tag=as51d966dd Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 150 ACK ACK Content-Length: 0 0 <-------------> --- (8 headers 0 lines) --- --- <--- SIP read from from UDP:192.168.1.204:5062 ---> INVITE ---> INVITE sip:310@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 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: "Tolstobrov A.Y." "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: <sip:310@192.168.1.100> Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 151 INVITE INVITE Contact: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.204:5062> Authorization: Digest username="110", username="110", realm="asterisk", nonce="7c3f36a8", uri="sip:310@192.168.1.100", response="cbd9f3618752a9c76c4ea3d23922377a", algorithm=MD5 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: "Tolstobrov A.Y." <sip:110@192.168.1.100> Supported: none P-Preferred-Identity: "User 110" <sip:110@192.168.1.100> Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 388 v=0 application/sdp, application/dtmf-relay Content-Length: 388 v=0 o=110 8002 8000 IN IP4 192.168.1.204 IP4 192.168.1.204 s=SIP Call Call c=IN IP4 192.168.1.204 192.168.1.204 t=0 0 0 m=audio 5008 RTP/AVP 0 8 4 18--- (18 headers 18 18 lines) --- --- Sending to to 192.168.1.204:5062 (NAT) (NAT) Using INVITE request as basis request - 678941221-5062-16@BJC.BGI.B.CAE Found - 678941221-5062-16@BJC.BGI.B.CAE Found peer '110' for '110' from 192.168.1.204:5062 from 192.168.1.204:5062 Found RTP audio format 0 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP 4 Found RTP audio format 18 18 Found RTP audio format 2 format 2 Found RTP audio format 97 Found RTP 97 Found RTP audio format 101 Found audio 101 Found audio description format PCMU for ID 0 Found 0 Found audio description format PCMA for ID 8 Found audio description format G723 G723 for ID 4 Found audio description 4 Found audio description format G729 for ID 18 Found audio 18 Found audio description format G726-32 for ID 2 Found audio description format iLBC iLBC for ID 97 Found audio description 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 - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined combined - 0x4 (ulaw) (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), (telephone-event|), peer - 0x1 (telephone-event|), (telephone-event|), combined - 0x1 (telephone-event|) Peer (telephone-event|) Peer audio RTP is at port 192.168.1.204:5008 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> hop: <sip:110@192.168.1.204:5062> <--- Transmitting (NAT) to to 192.168.1.204:5062 ---> ---> SIP/2.0 100 Trying Trying Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;received=192.168.1.204;rport=5062 From: "Tolstobrov A.Y." "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: <sip:310@192.168.1.100> Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 151 INVITE INVITE Server: Asterisk PBX 1.8.19.0 PBX 1.8.19.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: PUBLISH Supported: replaces, timer Contact: timer Contact: <sip:310@192.168.1.100:5060> Content-Length: 0 0 <------------> -- Executing [310@phones:1] Dial("SIP/110-00000168", "SIP/310") in in new stack -- Called SIP/310 -- SIP/310-00000169 is ringing ringing <--- Transmitting (NAT) to to 192.168.1.204:5062 ---> ---> SIP/2.0 180 Ringing Ringing Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;received=192.168.1.204;rport=5062 From: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 To: To: <sip:310@192.168.1.100>;tag=as5369cdd3 Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 151 INVITE INVITE Server: Asterisk PBX 1.8.19.0 PBX 1.8.19.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: PUBLISH Supported: replaces, timer Contact: timer Contact: <sip:310@192.168.1.100:5060> Content-Length: 0 0 <------------> -- SIP/310-00000169 answered SIP/110-00000168 SIP/110-00000168 Audio is at 13062 Adding codec 0x4 (ulaw) to SDP Adding SDP Adding non-codec 0x1 (telephone-event) to SDP SDP <--- Reliably Transmitting (NAT) to to 192.168.1.204:5062 ---> ---> SIP/2.0 200 OK OK Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK1991488353;received=192.168.1.204;rport=5062 From: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 To: To: <sip:310@192.168.1.100>;tag=as5369cdd3 Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 151 INVITE INVITE Server: Asterisk PBX 1.8.19.0 PBX 1.8.19.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: PUBLISH Supported: replaces, timer Contact: timer Contact: <sip:310@192.168.1.100:5060> Content-Type: application/sdp Content-Length: 266 v=0 266 v=0 o=root 1635771721 1635771721 IN IN IP4 192.168.1.100 192.168.1.100 s=Asterisk PBX 1.8.19.0 PBX 1.8.19.0 c=IN IP4 192.168.1.100 192.168.1.100 t=0 0 0 m=audio 13062 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 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 a=sendrecv <------------> -- Locally bridging SIP/110-00000168 and SIP/310-00000169 SIP/310-00000169 <--- SIP read from from UDP:192.168.1.204:5062 ---> ACK ---> ACK sip:310@192.168.1.100:5060 SIP/2.0 Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.204:5062;branch=z9hG4bK623808183;rport From: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 To: To: <sip:310@192.168.1.100>;tag=as5369cdd3 Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 151 ACK Contact: ACK Contact: <sip:110@192.168.1.204:5062> Max-Forwards: 70 70 Supported: replaces, replaces, path, timer, eventlist User-Agent: eventlist User-Agent: Grandstream DP715 1.0.0.23 Allow: 1.0.0.23 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 0 <-------------> --- (12 headers 0 lines) --- --- == Spawn extension (phones, 310, 1) 1) exited non-zero on 'SIP/110-00000168' Scheduling destruction of SIP dialog dialog '678941221-5062-16@BJC.BGI.B.CAE' in in 32000 ms (Method: ACK) setdestination: Parsing set_destination: Parsing <sip:110@192.168.1.204:5062> for for address/port to send to setdestination: set_destination: set destination to 192.168.1.204:5062 to 192.168.1.204:5062 Reliably Transmitting (NAT) to 192.168.1.204:5062: to 192.168.1.204:5062: BYE sip:110@192.168.1.204:5062 SIP/2.0 Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK766d9793;rport Max-Forwards: 70 From: 70 From: <sip:310@192.168.1.100>;tag=as5369cdd3 To: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 102 BYE BYE User-Agent: Asterisk PBX 1.8.19.0 PBX 1.8.19.0 Proxy-Authorization: Digest username="110", realm="asterisk", algorithm=MD5, realm="asterisk", algorithm=MD5, uri="sip:192.168.1.100", nonce="", nonce="", response="e4f3d1f63862a146ddcb56ddbc147491" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 0 --- <--- SIP read from from UDP:192.168.1.204:5062 ---> SIP/2.0 ---> SIP/2.0 200 OK OK Via: SIP/2.0/UDP SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK766d9793;rport=5060 From: From: <sip:310@192.168.1.100>;tag=as5369cdd3 To: "Tolstobrov A.Y." "User 110" <sip:110@192.168.1.100>;tag=1842811006 Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: Call-ID: 678941221-5062-16@BJC.BGI.B.CAE CSeq: 102 BYE Contact: BYE Contact: <sip:110@192.168.1.204:5062> Supported: replaces, path, timer, eventlist timer, eventlist User-Agent: Grandstream Grandstream DP715 1.0.0.23 1.0.0.23 Allow: INVITE, ACK, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 0 <-------------> --- (11 headers 0 lines) --- --- SIP Response message for INCOMING dialog dialog BYE arrived arrived Really destroying SIP dialog '678941221-5062-16@BJC.BGI.B.CAE' SIP dialog '678941221-5062-16@BJC.BGI.B.CAE' Method: ACK

ACK

История изменений [назад]

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!

Asterisk + IP Dect Phones Grandstream DP710/715 ЭХО!