Asterisk stop detecting the SIP line and needs a restart

I am facing continuous issue with asterisk where asterisk stops responding to the incoming calls. The end user(Caller) get a busy tone in this case. On restarting asterisk the line starts working fine.

I am unable to detect when this happens and issue is generally raised by the client which is not obviously not right.

This issue could be due to the SIP line bouncing but i am not very sure of it.

Can we detect when the SIP line stops working? Also can we handle this automatically without manual intervention?

What do you mean by “SIP Line Bouncing”?

If qualify is enabled, Asterisk will probe the communications to each peer and mark them unreachable.

Really, though, you need to provide provide sufficient debugging information to see what is happening. At the very least run with “sip set debug on”.

This could well be a deadlock, for which you need to follow the instructions in wiki.asterisk.org/wiki/display/ … rADeadlock

Thanks David.

In case the SIP line goes down. When the line is fixed I need to restart asterisk otherwise asterisk does not respond to the incoming calls at. Asterisk behaves as if its not receiving any signal at all.

That is the reason I cannot provide you any log from “sip set debug on”.

If I restart asterisk the line starts working fine.

That sounds like a deadlock. However, you can still provide the SIP debugging from immediately before it goes wrong.

However, as you are not completely clear about what is going on there is a slight possibility that a firewall or NAT rule has timed out.

This might help

asterisk-rd.blogspot.com/2013/04 … o-sip.html

issues.asterisk.org/jira/browse/ASTERISK-18930

Restarting Asterisk won’t clear a DNS problem.

It hasn’t been told in none of the Articles that restarting Asterisk , could clear a DNS problem. It just makes reference that a DNS problem could be the cause why SIP, stop responding