The above calls was actually not answered which was suppose to received the dial status as No ANSWER but instead of that am getting as BUSY
-- SIP/Operator-sip-trunk-0000004f is busy
Scheduling destruction of SIP dialog ‘79b71c8808780032471312480fe7a923@10.10.10.2:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:1/0/0)
Is that anything am doing wrong here or how can I get the proper dialstatus for this call.
There is only one call in the log, so I’m not sure which of your cases it represents.
It includes 181 Call is being forwarded, followed by 400 service unavailable, so my guess is this is the no answer case and it was timed out by the B side, and sent to voicemail, which failed, before Asterisk timed it out.
You haven’t specified a timeout when dialling, and I don’t know if that is even possible from the console. Without a timeout, you won’t get a NO ANSWER status.
I have not specified the timeout because I dont wanted to get the call disconnect from the Asterisk end and the BYE packet needed from the Telecom operators.
On the above, the calls were initiated from Console directly to Phonenumber through telecom operators SIP trunk, and the mobile number who got the calls was not answered.
The only way that calls can be detected as not answered is by a time out. There is no way for SIP to signal that the end user didn’t answer, and, as I said in this case, it looks like it actually signalled a transfer followed by a general failure.
The nearest SIP status is intended for timeouts due to technical failures, not those due to users not picking up the phone.