Asterisk terminating outbound call when picked up, sends 'BYE' message


#1

HAII…

I have installed asterisk 14 .

And am using sip trunk to make the call.

Asterisk terminating outbound call when picked up, sends ‘BYE’ message.

Below is the Enable debug message.

Can anyone help on this?


#2

Yes, that is a log of a normal BYE transaction.

I don’t understand what you think is wrong and you have provided no evidence of a problem.


#3

I am dialing the number from the asterisk server using voip ,And i am facing the issue as some times when end user answer the call the call is getting disconnecting in 2 to 5 seconds .

As i have checked the BYE is sending from the asterisk server.

How can avoid this call disconnection .


#4

By finding and removing the cause. You have not provided any information that would allow us to do so. The reason is likely to be in the Asterisk logs.


#5

Hai

Please find the below log and suggest.

– (11 headers 10 lines) —
sip_route_dump: route/path hop: sip:callee@108.59.2.134;did=951.1832f504
Transmitting (NAT) to 108.59.2.134:5060:
ACK sip:callee@108.59.2.134;did=951.1832f504 SIP/2.0
Via: SIP/2.0/UDP 172.31.31.111:5061;branch=z9hG4bK3f018da0;rport
Max-Forwards: 70
From: sip:+917574832699@34.208.153.217:5061;tag=as0cda9c2b
To: sip:0011102919739962916@sbc.voip.com;tag=3737942390-1510275050
Contact: sip:+917574832699@172.31.31.111:5061
Call-ID: 67374d933dab14c73d67b5c02a44808e@34.208.153.217
CSeq: 102 ACK
User-Agent: FPBX-14.0.3.2(14.7.6)
Content-Length: 0


-- SIP/voip_out-000000ce answered
-- Executing [s@FLOW_EXECUTOR_DAILPLAN:1] Set("SIP/voip_out-000000ce", "CDR(Host)=pbx-1") in new stack
-- Executing [s@FLOW_EXECUTOR_DAILPLAN:2] NoOp("SIP/voip_out-000000ce", "pbx-1, 1528953589.206 ") in new stack
-- Executing [s@FLOW_EXECUTOR_DAILPLAN:3] Wait("SIP/voip_out-000000ce", "0.25") in new stack
-- Manager 'admin' from 127.0.0.1, hanging up channel: SIP/voip_out-000000ce

== Spawn extension (FLOW_EXECUTOR_DAILPLAN, s, 3) exited non-zero on ‘SIP/voip_out-000000ce’
Scheduling destruction of SIP dialog ‘67374d933dab14c73d67b5c02a44808e@34.208.153.217’ in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 108.59.2.134:5060:
BYE sip:callee@108.59.2.134;did=951.1832f504 SIP/2.0
Via: SIP/2.0/UDP 172.31.31.111:5061;branch=z9hG4bK272f27e4;rport
Max-Forwards: 70
From: sip:+917574832699@34.208.153.217:5061;tag=as0cda9c2b
To: sip:0011102919739962916@sbc.voip.com;tag=3737942390-1510275050
Call-ID: 67374d933dab14c73d67b5c02a44808e@34.208.153.217
CSeq: 103 BYE
User-Agent: FPBX-14.0.3.2(14.7.6)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from UDP:108.59.2.134:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.31.111:5061;received=34.208.153.217;rport=5061;branch=z9hG4bK272f27e4
To: sip:0011102919739962916@sbc.voip.com;tag=3737942390-1510275050
From: sip:+917574832699@34.208.153.217;tag=as0cda9c2b
Call-ID: 67374d933dab14c73d67b5c02a44808e@34.208.153.217
CSeq: 103 BYE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Reason: Q.850;cause=16;text=“Normal call clearing”
Content-Length: 0


#6

AMI was used to terminate the call.


#7

If you read and analyze carefully your logs you will know who and why the call is terminated