Call failed to go through, reason (3) Remote end Ringing


I use call files to announce alarms. A call should be done and after Answer a voice file should be played. Most of the time everything happens as I like. Sometimes I get the error in topic. The asterisk server is connected to FritzBox as SIP Client

Asterisk version: 16.28.0~dfsg-0+deb11u3


Channel: SIP/01xxxxx@fb_tom
Callerid: 08xxxxx
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: von_datei
Extension: 10


register => xxx:xxx@192.168.x.x/xxx



exten => 10,1,MixMonitor(${UNIQUEID}.wav)
exten => 10,n,Answer()
exten => 10,n,Wait(1)
exten => 10,n,Playback(/home/test/audios/unter4)
exten => 10,n,Wait(1)
exten => 10,n,Hangup()

For tests a called my own cellphone several times. In 9 of 10 tries everything works out fine. But in one I get the error.
Maybe someone has an Idea, what to do?

Thanks a lot

Cause code 3 is “no route to destination”. Please provide the full log line containing the message, as I don’t think is an Asterisk one. Actually provide the full log for the whole call attempt, with verbosity at least 3.

Your context names don’t match. This should never work.

chan_sip is unsupported, and no longer in the source code.

Asterisk 16 is unsupported.

Asterisk 16.28.0 has known security vulnerabilities.

defaultuser is of no use when not used with host=dynamic, is an obsolete parameter name, and is not normally needed, evern when meaningful.

Hi David,

thanks for the fast answer. I’m sorry, I missed to catch the log. Now I put verbosity to 3 and did several test calls. Everything works fine. So I can’t provide log data right now. I’ll try on.
Why do you think I have context mismatch? I did everything copy paste from the files.

I’m not able to compile a new Asterisk by my self, therefor my linux skills are to bad. Do you know how I can easy get a current version? I did installation of Debian 11 and then install Asterisk with apt.

Thanks a lat


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