#PJSIP #Asterisk22 Asterisk failes to send BYE-Request when user hangs up

Asterisk Asterisk SIP
1

Dear Asterisk-Community,

when the user beyond the Asterisk 22 receives a call and hangs up after the call Asterisk doesn’t send a BYE-Request. Consequence is that the call isn’t hang up on the remote party. On the remote party the call remains but silent.
This is the inbound section of the extensions.conf:

[from-giganetz]
exten => +49xxx,1,Dial(PJSIP/one,20,rt)
same => n,VoiceMail(mbone@default)
same => n,Hangup

The hardware beyond the Asterisk is a Gigaset-Go-System; the remote party is a Samsung cell phone registered in the Telefonica 4G network. SIP provider is Deutsche GigaNetz GmbH who uses Purtel’s infrastructure; the subscription is MyNet 600 with three landline numbers. The Asterisk runs on a Debian stable VM.
This is the SIP trace in Wireshark:

SIP trace
SIP trace1047×114 13.8 KB

The fault is reproducable. I tested it with other hardware which isn’t based on Asterisk. It sends the BYE-Request and the fault doesn’t appear.
What have I to do in order to fix the fault?
Thank you very much!

Bes regards

Jung-Fernmelder

2

The remote party is the one that is responsible for sending the BYE request.

Your log (which really should be provided as plain text) shows a successful outgoing SIP call that hasn’t been ended by either side. It shows no indication of any event that might cause Asterisk to want to end it.

Please enable the Asterisk full log, and use “pjsip set logger on”, and provide a log containing an event that you believe should have caused Asterisk to initiate the ending of the outbound leg.

Note that a reason for failing to receivn an incoming BYE request might be that you have set the wrong external signalling address on the transport. For that reason,please also include, at least, the relevant transport and other pjsip.conf sections. relating to the outbound endpoint.

I’m assuming that the endpoint is outside NAT based on the fact that you felt it necessary to redact its address.