Calls do not drop via asterisk user

hi is there a way to investigate why calls made via asterisk users are not dropped when i drop the call on my softphone, call is still on on my mobile i have to close it manually.

What channel technology are you using for the outgoing connection?

I assume softphone implies SIP; which of chan_sip or chan_pjsip are you using?

I’m not aware of any mechanism by which a call originated by SIP, or analogue FXS and completed by SIP, ISDN, or analogue FXO, would fail to clear promptly. You can expect problems in clearing incoming analogue FXO calls.

To investigate, you need to work out whether the problem is on the incoming or the outgoing side, then enable protocol debugging for that side.

im using chan_sip and the issue is only on the outgoing side.

Issue “sip set debug on” using the CLI, and look at the full log to confirm that a BYE is being sent and an OK is being received.

If both are present, the problem is out of your control, as it is with the provider.

If BYE is being sent and no ACK being received, you may have a NAT configuration problem.

If BYE isn’t being sent, I would say the problem was on the incoming side. but confirm that no BYE is being received.

If BYE is being received but not sent, I think we will need the complete contents of the full log from the receipt of the BYE until it is certain that none is going to be sent.

If providing logs, sensitive information should be removed, but in a way that preserves the distinction between different addresses and between public and private addresses, the log uploaded to and a link provide here.

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