I can call to unavaliable extension

Hi, I have a small issue.

I have number 304 which is unavailable, and when i call to this number from another extension i have a call… my headset ringing but where? How to change this and when number is unavailable i want instant hangup.

asterisk -rvvv will likely tell you which devices are affected.

Define unavailable. Define extension.

I think, by extension, you mean a SIP device, not an Asterisk extension, and by unavailable, I guess you mean doesn’t produce any response to SIP INVITEs.

Given those assumptions, which SIP channel driver are you using, the officulaly supported chan_pjisp, or the legacy chan_sip, which only has community support?

SIP operates over unreliable networks, with finite propagation delays, so it is impossible to get an instantaneous response (even ignoring the speed of light absolute limit on responses, and the failure to get a response to the first attempt doesn’t mean the call will fail. Also, the called system is allowed a certain grace period before sending even a provisional response.

Consequently, SIP makes several attempts to set up the call, which, with default parameters take about 30 seconds to complete.

If you know that a destination has a low worst case round trip delay, or is highly reliable, you can modify the timeouts used for this purpose.

Also, asterisk has a feature in which it tests connectivity, by sending OPTIONS requests, and will mark a peer as down if it fails to respond within a specified time. As this has to poll, and also has, itself, to allow for lost packets, this will take a finite time to detect loss of connectivity. Some peers don’t play ball with this mechanism, and it has to be turned off for them.

If the destination has to register, it will have to periodically refresh the registration, and that is likely done less often than qualify.

If you are getting ringback tone, this is the result of how you wrote the dialplan (if you didn’t write the dialplan, you are on the wrong forum), or possibly how the originating phone works. Asterisk doesn’t send Ringing, or ringback tone, unless it receives Ringing from the far end, or the dialplan instructs it to do so.

Basically, the nature of VoIP networks means you can’t get speed of light confirmations of the state of the other party, but tweaking timeouts, may speed up detection of non-responding destinations, and tweaking qualify and registration timers may increase the chance of the loss of connectivity being detected ahead of the call, but any ringback tone associated with a non-responding destination is either not from Asterisk, or because someone configured Asterisk to send it.


You can run command asterisk -rvvvv and see what happen. Maybe you have followme enabled on this extension.

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