アスタリスクのコール終了

Question

プロバイダから番号をダイヤルしようとすると、応答の直後に接続が中断されます。つまり、同じ設定では通話が終了し、通話が切断されます。この振る舞いをどのような方向性につなげることができるのか？ SIP-コールログ：

m2422*CLI> channel originate SIP/<some number>@<provider's ip> application MusicOnHold 

    == Using SIP RTP CoS mark 5 
Audio is at 33966 
Adding codec ulaw to SDP 
Adding codec alaw to SDP 
Adding non-codec 0x1 (telephone-event) to SDP 
Reliably Transmitting (no NAT) to <provider's ip>:5060: 
INVITE sip:<some number>@<provider's ip> SIP/2.0 
Via: SIP/2.0/UDP <my ip>:5060;branch=z9hG4bK478c225f 
Max-Forwards: 70 
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
To: <sip:<some number>@<provider's ip>> 
Contact: <sip:anonymous@<my ip>:5060> 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
CSeq: 102 INVITE 
User-Agent: docker 
Date: Thu, 13 Apr 2017 21:39:31 GMT 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 
Supported: replaces, timer 
Content-Type: application/sdp 
Content-Length: 265 

v=0 
o=root 1062463446 1062463446 IN IP4 <my ip> 
s=Asterisk PBX 14.3.0 
c=IN IP4 <my ip> 
t=0 0 
m=audio 33966 RTP/AVP 0 8 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:8 PCMA/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=maxptime:150 
a=sendrecv 

--- 
    -- Called <some number>@<provider's ip> 

<--- SIP read from UDP:<provider's ip>:5060 ---> 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP <my ip>:5060;branch=z9hG4bK478c225f 
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
To: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
Date: Thu, 13 Apr 2017 21:39:31 GMT 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
Server: Cisco-SIPGateway/IOS-12.x 
CSeq: 102 INVITE 
Allow-Events: telephone-event 
Content-Length: 0 


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

<--- SIP read from UDP:<provider's ip>:5060 ---> 
SIP/2.0 183 Session Progress 
Via: SIP/2.0/UDP <my ip>:5060;branch=z9hG4bK478c225f 
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
To: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
Date: Thu, 13 Apr 2017 21:39:31 GMT 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
Server: Cisco-SIPGateway/IOS-12.x 
CSeq: 102 INVITE 
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER 
Allow-Events: telephone-event 
Contact: <sip:<some number>@<provider's ip>:5060> 
Content-Disposition: session;handling=required 
Content-Type: application/sdp 
Content-Length: 259 

v=0 
o=CiscoSystemsSIP-GW-UserAgent 7410 4097 IN IP4 <provider's ip> 
s=SIP Call 
c=IN IP4 <provider's ip> 
t=0 0 
m=audio 18808 RTP/AVP 0 101 
c=IN IP4 <provider's ip> 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=direction:passive 

<-------------> 
--- (14 headers 11 lines) --- 
sip_route_dump: route/path hop: <sip:<some number>@<provider's ip>:5060> 
Found RTP audio format 0 
Found RTP audio format 101 
Found audio description format PCMU for ID 0 
Found audio description format telephone-event for ID 101 
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) 
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) 
Peer audio RTP is at port <provider's ip>:18808 
    -- SIP/trunk-0000001b is making progress 
     > 0x7f75f8002870 -- Probation passed - setting RTP source address to <provider's ip>:18808 

<--- SIP read from UDP:<provider's ip>:5060 ---> 
SIP/2.0 183 Session Progress 
Via: SIP/2.0/UDP <my ip>:5060;branch=z9hG4bK478c225f 
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
To: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
Date: Thu, 13 Apr 2017 21:39:31 GMT 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
Server: Cisco-SIPGateway/IOS-12.x 
CSeq: 102 INVITE 
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER 
Allow-Events: telephone-event 
Contact: <sip:<some number>@<provider's ip>:5060> 
Content-Disposition: session;handling=required 
Content-Type: application/sdp 
Content-Length: 259 

v=0 
o=CiscoSystemsSIP-GW-UserAgent 7410 4097 IN IP4 <provider's ip> 
s=SIP Call 
c=IN IP4 <provider's ip> 
t=0 0 
m=audio 18808 RTP/AVP 0 101 
c=IN IP4 <provider's ip> 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=direction:passive 

<-------------> 
--- (14 headers 11 lines) --- 
sip_route_dump: route/path hop: <sip:<some number>@<provider's ip>:5060> 
    -- SIP/trunk-0000001b is making progress 

<--- SIP read from UDP:<provider's ip>:5060 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP <my ip>:5060;branch=z9hG4bK478c225f 
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
To: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
Date: Thu, 13 Apr 2017 21:39:31 GMT 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
Server: Cisco-SIPGateway/IOS-12.x 
CSeq: 102 INVITE 
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER 
Supported: replaces 
Allow-Events: telephone-event 
Contact: <sip:<some number>@<provider's ip>:5060> 
Content-Type: application/sdp 
Content-Length: 259 

v=0 
o=CiscoSystemsSIP-GW-UserAgent 7410 4097 IN IP4 <provider's ip> 
s=SIP Call 
c=IN IP4 <provider's ip>5 
t=0 0 
m=audio 18808 RTP/AVP 0 101 
c=IN IP4 <provider's ip> 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=direction:passive 

<-------------> 
--- (14 headers 11 lines) --- 
sip_route_dump: route/path hop: <sip:<somenumber>@<provider's ip>:5060> 
set_destination: Parsing <sip:<somenumber>@<provider's ip>:5060> for address/port to send to 
set_destination: set destination to <provider's ip>:5060 
Transmitting (no NAT) to <provider's ip>:5060: 
ACK sip:<some number>@<provider's ip>:5060 SIP/2.0 
Via: SIP/2.0/UDP <my ip>:5060;branch=z9hG4bK2566cc60 
Max-Forwards: 70 
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
To: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
Contact: <sip:anonymous@<my ip>:5060> 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
CSeq: 102 ACK 
User-Agent: docker 
Content-Length: 0 


--- 
    -- SIP/trunk-0000001b answered 
     > Launching MusicOnHold() on SIP/trunk-0000001b 
    -- Started music on hold, class 'default', on channel 'SIP/trunk-0000001b' 

<--- SIP read from UDP:<provider's ip>:5060 ---> 
BYE sip:anonymous@<my ip>:5060 SIP/2.0 
Via: SIP/2.0/UDP <provider's ip>:5060;branch=z9hG4bK67DF6A22C1 
From: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
Date: Thu, 13 Apr 2017 21:39:42 GMT 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
User-Agent: Cisco-SIPGateway/IOS-12.x 
Max-Forwards: 70 
Timestamp: 1492119582 
CSeq: 101 BYE 
Reason: Q.850;cause=16 
Content-Length: 0 


<-------------> 
--- (12 headers 0 lines) --- 
Sending to <provider's ip>:5060 (no NAT) 
Scheduling destruction of SIP dialog '264825d83272bc8d676c07b27e9cb754@<my ip>:5060' in 32000 ms (Method: BYE) 

<--- Transmitting (no NAT) to <provider's ip>:5060 ---> 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP <provider's ip>:5060;branch=z9hG4bK67DF6A22C1;received=<provider's ip> 
From: <sip:<some number>@<provider's ip>>;tag=488081DC-9C3 
To: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as37dc79d9 
Call-ID: 264825d83272bc8d676c07b27e9cb754@<my ip>:5060 
CSeq: 101 BYE 
Server: docker 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE 
Supported: replaces, timer 
Content-Length: 0 


<------------> 
    -- Stopped music on hold on SIP/trunk-0000001b 

Answer 1

<--- SIP read from UDP:<provider's ip>:5060 ---> 
BYE sip:anonymous@<my ip>:5060 SIP/2.0 

あなた持っている問題

あるものを提供尋ねる

ほとんどlikly - プロバイダはお勧めしませあなたの側から無音（間違ったNATとexternip設定）か、コーデックを使用/サポートしません。