Hi All
AP200d настроен в режиме выноски абонентской линии, он же hotline,протокол sip.
Оба порта FXO заходят в атс
При входящем звонке AP сразу даёт сигнал на порт, атс поднимает трубку, звонящий слышет сигнал и донабирает номер.
Проблема в том, что если сразу же снять трубку то звонящего не слышно 5-6 сек. Почему так и не понял, но наблюдается это только если с другой стороны linksys. Ну да это всё лирика, что то мне подсказывает, что если AP будет вызывное подавать в порт не сразу а через 2-3 сек - будет гораздо проще. Вот только не знаю как этот таймер называется, help. Т.е. я хочу добиться след поведения - при входящем AP ждёт 2-3 сек, потом подаёт сигнал на станцию, та поднимает трубку, ждёт донабора номера. Вот как то так.
Впрочем, лог ниже. Вдруг кто увидит ошибку опытным глазом
Received SIP PDU from ( 10.17.5.245:5060 )
INVITE sip:189@10.17.2.4 SIP/2.0
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To:
Remote-Party-ID: 2 ;screen=yes;party=calling
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 2
Expires: 240
User-Agent: Linksys/SPA2102-5.1.9
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 59804141 59804141 IN IP4 10.17.5.245
s=-
c=IN IP4 10.17.5.245
t=0 0
m=audio 16406 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To:
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0
1 : ****************** Call Created status(InitiatedByNet) *******************
2 : Receive INVITE Request
3 : Found inbound voip peer by dest-pattern id(2)
4 : From Net - calledParty(189) callingParty(2)
5 : MatchedPerfect
6 : MatchAllProcess After Sorted
id(20) dest(189) prefer(0) selected(0)
7 : Initiate callee with dial-peer(189) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
8 : InitiateOutCall : calledNum(), callingNum(2), callerPort(ffffffff) type(FXO)
[479.595] RTA(0/1/0) Rx CC_OFFHOOK_REQ peerId(-1)
[479.595] VM(0/1/0) FXO OffHook
[479.595] VM(0/1/0) vopp enable
[479.595] VM(0/1/0) Fax enable
[479.595] VM(0/1/0) play mute
[479.600] VM(0/1/0) Tx CONNECT_CNF
9 : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(2)
[479.600] VM(0/1/0) set T38 enable
[479.600] VM(0/1/0) set T38 mode STD
[479.600] VM(0/1/0) Fax rate 9600
10 : SetAlerting
11 : PreConnected from(100)
[479.600] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[479.600] VM(0/1/0) VAD disable
[479.600] VM(0/1/0) SID enable by CCC
12 : Add Local Audio MediaFormat : 0
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:189@10.17.2.4
Content-Type: application/sdp
Content-Length: 182
v=0
o=189 1277720553 1277720553 IN IP4 10.17.2.4
s=AddPac Gateway SDP
c=IN IP4 10.17.2.4
t=1277720553 0
m=audio 23004 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=sendrecv
[479.630] RTA(0/1/0) Rx RS_OPEN_REQ callId=2 ssId=1 G711U
peer=10.17.5.245 mp=23004/23005 hp=16406/16407
[479.630] VM(0/1/0) vopp idle
[479.630] VM(0/1/0) start codec replace timer to G711U
13 : Connected from(100)
[479.640] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[479.640] VM(0/1/0) VAD disable
[479.640] VM(0/1/0) SID enable by CCC
14 : SetConnected
15 : Add Local Audio MediaFormat : 0
[479.660] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:189@10.17.2.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 182
v=0
o=189 1277720553 1277720553 IN IP4 10.17.2.4
s=AddPac Gateway SDP
c=IN IP4 10.17.2.4
t=1277720553 0
m=audio 23004 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=sendrecv
[479.670] RTA(0/1/0) Rx RS_LISTEN_REQ callId=2 ssId=1 G711U
peer=10.17.5.245 mp=23004/23005 hp=16406/16407
[479.670] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 0
[479.670] VM(0/1/0) DTMF disable
Received SIP PDU from ( 10.17.5.245:5060 )
ACK sip:189@10.17.2.4 SIP/2.0
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-62faf134
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 ACK
Max-Forwards: 70
Contact: 2
User-Agent: Linksys/SPA2102-5.1.9
Content-Length: 0
[479.690] VM(0/1/0) vopp enable
[479.690] VM(0/1/0) codec replaced to G711U
[479.690] VM(0/1/0) Fax enable
[479.690] VM(0/1/0) play mute
16 : ACK received
17 : Receive ACK Request
18 : Set Terminated Success for 101 INVITE
Received SIP PDU from ( 10.17.5.245:5060 )
BYE sip:189@10.17.2.4 SIP/2.0
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-c94a6655
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 102 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA2102-5.1.9
Content-Length: 0
19 : Receive BYE Request
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-c94a6655
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 102 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0
20 : ReleaseWithNothing
[503.970] RTA(0/1/0) Rx RS_CLOSE_REQ callId=2 ssId=1 dir=reve
[503.970] RTA(0/1/0) Rx RS_CLOSE_REQ callId=2 ssId=1 dir=forw
[503.970] RTA(0/1/0) close Media socket
[503.970] RTA(0/1/0) close RTCP socket
21 : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0)
22 : DisconnectCall at Busy
23 : StopSignal
[503.975] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[503.975] VM(0/1/0) play mute
24 : Disconnect (0)
[503.975] RTA(0/1/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[503.975] VM(0/1/0) vopp idle
[503.975] VM(0/1/0) FXO OnHook
[503.975] VM(0/1/0) Tx DISCONN_CNF
25 : Call TO terminated reason(Remote:CallClear)
26 : Disconnected(16) at Disconnecting
27 : Set Terminated Success for 102 BYE
AP200d настроен в режиме выноски абонентской линии, он же hotline,протокол sip.
Оба порта FXO заходят в атс
При входящем звонке AP сразу даёт сигнал на порт, атс поднимает трубку, звонящий слышет сигнал и донабирает номер.
Проблема в том, что если сразу же снять трубку то звонящего не слышно 5-6 сек. Почему так и не понял, но наблюдается это только если с другой стороны linksys. Ну да это всё лирика, что то мне подсказывает, что если AP будет вызывное подавать в порт не сразу а через 2-3 сек - будет гораздо проще. Вот только не знаю как этот таймер называется, help. Т.е. я хочу добиться след поведения - при входящем AP ждёт 2-3 сек, потом подаёт сигнал на станцию, та поднимает трубку, ждёт донабора номера. Вот как то так.
Впрочем, лог ниже. Вдруг кто увидит ошибку опытным глазом
Received SIP PDU from ( 10.17.5.245:5060 )
INVITE sip:189@10.17.2.4 SIP/2.0
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To:
Remote-Party-ID: 2 ;screen=yes;party=calling
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 2
Expires: 240
User-Agent: Linksys/SPA2102-5.1.9
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 59804141 59804141 IN IP4 10.17.5.245
s=-
c=IN IP4 10.17.5.245
t=0 0
m=audio 16406 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To:
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0
1 : ****************** Call Created status(InitiatedByNet) *******************
2 : Receive INVITE Request
3 : Found inbound voip peer by dest-pattern id(2)
4 : From Net - calledParty(189) callingParty(2)
5 : MatchedPerfect
6 : MatchAllProcess After Sorted
id(20) dest(189) prefer(0) selected(0)
7 : Initiate callee with dial-peer(189) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
8 : InitiateOutCall : calledNum(), callingNum(2), callerPort(ffffffff) type(FXO)
[479.595] RTA(0/1/0) Rx CC_OFFHOOK_REQ peerId(-1)
[479.595] VM(0/1/0) FXO OffHook
[479.595] VM(0/1/0) vopp enable
[479.595] VM(0/1/0) Fax enable
[479.595] VM(0/1/0) play mute
[479.600] VM(0/1/0) Tx CONNECT_CNF
9 : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(2)
[479.600] VM(0/1/0) set T38 enable
[479.600] VM(0/1/0) set T38 mode STD
[479.600] VM(0/1/0) Fax rate 9600
10 : SetAlerting
11 : PreConnected from(100)
[479.600] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[479.600] VM(0/1/0) VAD disable
[479.600] VM(0/1/0) SID enable by CCC
12 : Add Local Audio MediaFormat : 0
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:189@10.17.2.4
Content-Type: application/sdp
Content-Length: 182
v=0
o=189 1277720553 1277720553 IN IP4 10.17.2.4
s=AddPac Gateway SDP
c=IN IP4 10.17.2.4
t=1277720553 0
m=audio 23004 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=sendrecv
[479.630] RTA(0/1/0) Rx RS_OPEN_REQ callId=2 ssId=1 G711U
peer=10.17.5.245 mp=23004/23005 hp=16406/16407
[479.630] VM(0/1/0) vopp idle
[479.630] VM(0/1/0) start codec replace timer to G711U
13 : Connected from(100)
[479.640] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[479.640] VM(0/1/0) VAD disable
[479.640] VM(0/1/0) SID enable by CCC
14 : SetConnected
15 : Add Local Audio MediaFormat : 0
[479.660] RTA: rtaMsgRxHandle RTP recvfrom on Forw session
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-f1e192b7
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:189@10.17.2.4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 182
v=0
o=189 1277720553 1277720553 IN IP4 10.17.2.4
s=AddPac Gateway SDP
c=IN IP4 10.17.2.4
t=1277720553 0
m=audio 23004 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=ptime:20
a=sendrecv
[479.670] RTA(0/1/0) Rx RS_LISTEN_REQ callId=2 ssId=1 G711U
peer=10.17.5.245 mp=23004/23005 hp=16406/16407
[479.670] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 0
[479.670] VM(0/1/0) DTMF disable
Received SIP PDU from ( 10.17.5.245:5060 )
ACK sip:189@10.17.2.4 SIP/2.0
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-62faf134
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 101 ACK
Max-Forwards: 70
Contact: 2
User-Agent: Linksys/SPA2102-5.1.9
Content-Length: 0
[479.690] VM(0/1/0) vopp enable
[479.690] VM(0/1/0) codec replaced to G711U
[479.690] VM(0/1/0) Fax enable
[479.690] VM(0/1/0) play mute
16 : ACK received
17 : Receive ACK Request
18 : Set Terminated Success for 101 INVITE
Received SIP PDU from ( 10.17.5.245:5060 )
BYE sip:189@10.17.2.4 SIP/2.0
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-c94a6655
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 102 BYE
Max-Forwards: 70
User-Agent: Linksys/SPA2102-5.1.9
Content-Length: 0
19 : Receive BYE Request
Sending SIP PDU to ( 10.17.5.245:5060 ) from 5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.17.5.245:5060;branch=z9hG4bK-c94a6655
From: 2 ;tag=b10df1d9f7ec4c67o0
To: ;tag=e94c5e01a4
Call-ID: 73500c41-35b6b51f@10.17.5.245
CSeq: 102 BYE
User-Agent: AddPac SIP Gateway
Content-Length: 0
20 : ReleaseWithNothing
[503.970] RTA(0/1/0) Rx RS_CLOSE_REQ callId=2 ssId=1 dir=reve
[503.970] RTA(0/1/0) Rx RS_CLOSE_REQ callId=2 ssId=1 dir=forw
[503.970] RTA(0/1/0) close Media socket
[503.970] RTA(0/1/0) close RTCP socket
21 : Terminated from(fffffffe) this(Remote:CallClear) before(NULL) forced(0)
22 : DisconnectCall at Busy
23 : StopSignal
[503.975] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[503.975] VM(0/1/0) play mute
24 : Disconnect (0)
[503.975] RTA(0/1/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[503.975] VM(0/1/0) vopp idle
[503.975] VM(0/1/0) FXO OnHook
[503.975] VM(0/1/0) Tx DISCONN_CNF
25 : Call TO terminated reason(Remote:CallClear)
26 : Disconnected(16) at Disconnecting
27 : Set Terminated Success for 102 BYE