SOLVED Random external calls mute automatically after 15 minutes

[vchohan]

Hi Team, we are facing a strange issue with external calls, random calls are going to silent/mute after 15 minutes and there is no issue if we dial the same number again. Earlier, it was happening rarily but now, it is happening frequently.

Session timer=refuse
nat=yes
RTP range=10002 - 20999
NAT Mode= Yes- (Force-rport,comedia)

 

[krzykat]

Under Settings->Asterisk SIP Settings->Chan-SIP ... at the bottom you should have

Under Other SIP Settings:

session-timers=refuse

I note you said Session timer=refuse, and want to make sure you have the syntax properly. But that is the main place for this type of issue. That or a really shitty provider.

 

[vchohan]

Under Settings->Asterisk SIP Settings->Chan-SIP ... at the bottom you should have

Under Other SIP Settings:

session-timers=refuse

I note you said Session timer=refuse, and want to make sure you have the syntax properly. But that is the main place for this type of issue. That or a really shitty provider.

It was typo here actually session-timers=refuse mentioned under Other SIP Settings. Is there anything else I can check?

 

[atsak]

What kind of firewall do you have in front of the PBX or the phones?

 

[vchohan]

What kind of firewall do you have in front of the PBX or the phones?

For PBX, we are only using IP tables, however, phones are behind Sophos UTM firewall. But, we have allowed all RTP range and disbaled SIP ALG.

 

[atsak]

My guess is you still have a firewall problem here, you could run a packet trace and see if you see the RTP packets getting lost somewhere.
Another indirect way to do it would be to hook the phone up to a temporary internet connection or secondary IP on a different router (cheap consumer type one) and see if the problem goes away. Make sure it's public IP not private.
The other problem could be that fail2ban is getting an invalid message from your phone for some reason and is banning the IP.
The last problem I can think of is it could be that the carrier is moving the RTP proxy, in this case what I would do is change canreinvite=no and I think maybe sendrpid=no as well in the extension / trunk configuration and see if that makes a difference. This isn't very likely.

 

[dhoppy]

I'm no expert here, but I've fixed a similar problem by setting, on the phone (not the server), NAT to keep-alive. Guess it depends on the type of phone you are dealing with.

 

[vchohan]

Issue Fixed by adding local network range under Asterisk SIP setting. We are still monitoring.