尝试通过提供商拨打号码时，接听后立即中断。也就是说，在相同的设置下，呼叫通过，然后中断。这种行为可以与什么联系起来，寻找什么方向？SIP 通话记录：
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
<--- SIP read from UDP:<provider's ip>:5060 --->
BYE sip:anonymous@<my ip>:5060 SIP/2.0
你有问提供什么是问题
最有可能 - 您这边没有声音(不正确的nat和externip设置(或者您使用提供商不推荐/不支持的编解码器。