Не детектируется звук при ответе на входящий звонок
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Не детектируется звук при ответе на входящий звонок
Т.е. в случае если трубка снята вызывается фукция NVBackgroundDetect для детектирования продолжительности голосового ответа. Она должна детектировать продолжителность разговора в милисекундах (например для 'Алло' это где-то порядка 1620). Но на самом деле факт разговора не детектируется вообще - фукция сообщает что была тишина, хотя на самом деле реально на звонок ответили (не молчали).
Ниже привожу логи Asterisk-a:
asterisk*CLI>
We're at 192.168.20.8 port 63840
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 71.16.179.176:5060:
INVITE sip:427335879157813872@71.16.179.176 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK64f7c1af;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>
Contact: <sip:9157813872@192.168.20.8>
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Jul 2006 02:31:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 32028 32028 IN IP4 192.168.20.8
s=session
c=IN IP4 192.168.20.8
t=0 0
m=audio 63840 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
asterisk*CLI>
<-- SIP read from 71.16.179.176:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK64f7c1af;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>;tag=940226AF-BFA
Date: Wed, 05 Jul 2006 14:23:05 GMT
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines)---
asterisk*CLI>
<-- SIP read from 71.16.179.176:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK64f7c1af;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>;tag=940226AF-BFA
Date: Wed, 05 Jul 2006 14:23:05 GMT
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Contact: <sip:427335879157813872@71.16.179.176:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 181
v=0
o=CiscoSystemsSIP-GW-UserAgent 4732 823 IN IP4 71.16.179.176
s=SIP Call
c=IN IP4 71.16.179.176
t=0 0
m=audio 17782 RTP/AVP 0
c=IN IP4 71.16.179.176
a=rtpmap:0 PCMU/8000
--- (13 headers 8 lines)---
Found RTP audio format 0
Peer audio RTP is at port 71.16.179.176:17782
Found description format PCMU
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
asterisk*CLI>
<-- SIP read from 71.16.179.176:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK64f7c1af;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>;tag=940226AF-BFA
Date: Wed, 05 Jul 2006 14:23:05 GMT
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:427335879157813872@71.16.179.176:5060>
Content-Type: application/sdp
Content-Length: 181
v=0
o=CiscoSystemsSIP-GW-UserAgent 4732 823 IN IP4 71.16.179.176
s=SIP Call
c=IN IP4 71.16.179.176
t=0 0
m=audio 17782 RTP/AVP 0
c=IN IP4 71.16.179.176
a=rtpmap:0 PCMU/8000
--- (13 headers 8 lines)---
Found RTP audio format 0
Peer audio RTP is at port 71.16.179.176:17782
Found description format PCMU
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:427335879157813872@71.16.179.176:5060>
set_destination: Parsing <sip:427335879157813872@71.16.179.176:5060> for address/port to send to
set_destination: set destination to 71.16.179.176, port 5060
Transmitting (NAT) to 71.16.179.176:5060:
ACK sip:427335879157813872@71.16.179.176:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK4f4f402a;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>;tag=940226AF-BFA
Contact: <sip:9157813872@192.168.20.8>
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Starting SIP/71.16.179.176-a7aa at incoming,,1 failed so falling back to exten 's'
== Starting SIP/71.16.179.176-a7aa at incoming,s,1 still failed so falling back to context 'default'
-- Executing AGI("SIP/71.16.179.176-a7aa", "agi://192.168.20.9/voicecast") in new stack
-- AGI Script Executing Application: (NVBackgroundDetect) Options: (silence/5|x|250|500|2000|400)
-- Playing 'silence/5' (language 'en')
-- AGI Script Executing Application: (Background) Options: (/var/fourcreeks/55)
-- Playing '/var/fourcreeks/55' (language 'en')
== Spawn extension (default, s, 1) exited non-zero on 'SIP/71.16.179.176-a7aa'
set_destination: Parsing <sip:427335879157813872@71.16.179.176:5060> for address/port to send to
set_destination: set destination to 71.16.179.176, port 5060
Reliably Transmitting (NAT) to 71.16.179.176:5060:
BYE sip:427335879157813872@71.16.179.176:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK09a64179;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>;tag=940226AF-BFA
Contact: <sip:9157813872@192.168.20.8>
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 71.16.179.176:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.8:5060;branch=z9hG4bK09a64179;rport
From: "9157813872" <sip:9157813872@192.168.20.8>;tag=as72661fc1
To: <sip:427335879157813872@71.16.179.176>;tag=940226AF-BFA
Date: Wed, 05 Jul 2006 14:24:10 GMT
Call-ID: 07ed3c0a24b119e761bc4def22490d4f@192.168.20.8
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 103 BYE
--- (9 headers 0 lines)---
Destroying call '07ed3c0a24b119e761bc4def22490d4f@192.168.20.8'
Заранее признателен.
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