Asterisk 星号呼叫终止

当试图通过提供商拨打该号码时，应答后连接立即中断。也就是说，在相同的设置下，呼叫通过，然后中断。这一行为与什么有关，朝着什么方向寻找？SIP呼叫日志：

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

m2422*CLI>频道源SIP/@:5060:
邀请sip:@>
联系人：
电话号码：264825d83272bc8d676c07b27e9cb754@:5060
CSeq:102邀请
用户代理：docker
日期：2017年4月13日星期四格林威治时间21:39:31
允许：邀请、确认、取消、选项、再见、参考、订阅、通知、信息、发布、消息
支持：替换、定时器
内容类型：应用程序/sdp
内容长度：265
v=0
o=IP4中的根1062463446 1062463446
s=星号PBX 14.3.0
c=在IP4中
t=0
m=音频33966 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101电话事件/8000
a=fmtp:101 0-16
a=最大时间：150
a=sendrecv
---
--呼叫@:5060--->
SIP/2.0 100
Via:SIP/2.0/UDP:5060；分支=z9hG4bK478c225f
来自：“匿名”；标签=as37dc79d9
至：：5060--->
SIP/2.0 183会议进度
Via:SIP/2.0/UDP:5060；分支=z9hG4bK478c225f
来自：“匿名”；标签=as37dc79d9
至：：5060>
内容配置：会话；处理=必需
内容类型：应用程序/sdp
内容长度：259
v=0
o=IP4中的CiscoSystemsSIP GW用户代理7410 4097
t=0
m=音频18808 RTP/AVP 0 101
c=在IP4中：5060>
找到RTP音频格式0
找到RTP音频格式101
找到ID为0的音频描述格式PCMU
找到ID 101的音频描述格式电话事件
能力：美国-（ulaw | alaw），对等-音频=（ulaw）/视频=（无）/文本=（无），组合-（ulaw）
非编解码器功能（dtmf）：us-0x1（电话事件|）、对等-0x1（电话事件|）、组合-0x1（电话事件|）
对等音频RTP位于端口：18808
>;标签=488081DC-9C3
日期：2017年4月13日星期四格林威治时间21:39:31
电话号码：264825d83272bc8d676c07b27e9cb754@:5060
服务器：Cisco SIPGateway/IOS-12.x
CSeq:102邀请
允许：邀请、选项、再见、取消、确认、PRACK、COMET、参考、订阅、通知、信息、更新、注册
允许事件：电话事件
联系人：
s=SIP呼叫
c=在IP4中
a=rtpmap:0 PCMU/8000
a=rtpmap:101电话事件/8000
a=fmtp:101 0-16
a=方向：被动
---（14标题11行）---
sip_路由_转储：路由/路径跃点：：5060-->
SIP/2.0 200正常
Via:SIP/2.0/UDP:5060；分支=z9hG4bK478c225f
来自：“匿名”；标签=as37dc79d9
至：：5060>
内容类型：应用程序/sdp
内容长度：259
v=0
o=IP4 5中的CiscoSystemsSIP GW用户代理7410 4097
t=0
m=音频18808 RTP/AVP 0 101
c=在IP4中：5060>
set_目的地：解析：5060
传输（无NAT）至：5060 SIP/2.0
Via:SIP/2.0/UDP:5060；分支=z9hG4bK2566cc60
最大前锋：70
来自：“匿名”；标签=as37dc79d9
至：：5060--->
再见sip:anonymous@:5060 sip/2.0
Via:SIP/2.0/UDP>；标签=488081DC-9C3
致：“匿名”；标签=as37dc79d9
日期：2017年4月13日星期四格林威治时间21:39:42
电话号码：264825d83272bc8d676c07b27e9cb754@:5060
用户代理：Cisco SIPGateway/IOS-12.x
最大前锋：70
时间戳：1492119582
CSeq:101再见
原因：Q.850；原因=16
内容长度：0
---（12个标题0行）---
发送至：5060-->
SIP/2.0 200正常
通过：SIP/2.0/UDP
From:
Ok，您只需检查防火墙10000-20000560端口。这些端口已打开。当我接听电话时，我会听到大约1秒钟的声音（RTP已建立）。然后呼叫被终止。
<--- SIP read from UDP:<provider's ip>:5060 --->
BYE sip:anonymous@<my ip>:5060 SIP/2.0