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.
Could anybody tell me how I can fix this?
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.