Asterisk call termination - asterisk

Question

When trying to dial to the number through the provider, the connection is interrupted immediately after the answer. That is, on the same settings, the call goes through, then it breaks off. With what this behavior can be connected and in what direction to look for? SIP-call log:
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
You have ask provide what is issue
Most likly - no sound from your side(incorrect nat and externip settings) or you use codec which provider not recommend/not support.