I have a Fritzbox 7360 router. Since I’m using Asterisk, I configured it as a SIP-client on my Asterisk-server, so I can keep using my old analog phone. This seems to work fine, except for one little detail. If I hangup the horn, the call is not terminated. It just keeps going, until the other side hangs up as well.
If your PSTN connection is an analogue line, you will not be able to terminate the call from the called party end; you will have to wait for it to timeout. In the UK, this typically took three minutes, although it is being reduced to a few seconds.
This is a characteristic of the PSTN telphone service, designed to allow you to hang up one phone and pick up the call on a more convenient one.
You are right. When I initiate the call from the analog device, it will be terminated. But when it is answered from the analog device, it keeps running. I’ve been testing it up to 3+ minutes and it still didn’t terminate.
Is there anything I can do from my end to keep this under 3 minutes? Basically it isn’t my problem. But I would hate if someone would call me and forgets to hangup aferwards. Or simply trust I will terminate the call when I hangup. That could be an expensive mistake.