Issue : Incoming calls do not work after a time

Afternoon everyone,

I have an issue with my asterisk installation 13.21.1. I can make some oubound calls and receive calls. But after a time, any incoming calls do not work. I must to connect to the console CLI and enter the command “core restart now” before the incoming calls work again. Do you know where is the mater and how to fix it ?

Best regards,

No. If we did the way to fix it would be to install the latest version of Asterisk.

“Do not work” is a very imprecise statement. You need to be a lot more specific.

However, delayed failures can be the result of:

  • resource leaks;
  • deadlocks;
  • automatically generated rules in NAT routers and firewalls expiring;

Yes, when I run asterisk for the first time all work very well (oubound and inbound calls) but I have finally seen that my incoming calls do not work (The caller have the tonality but my sip phone don’t ring) after a time and in the CLI I have nothing when I look in the “pjsip set logger on” I have nothing too. But In the portal of my provider BelgiumVoIP in the history calls, I can see my differents incoming calls display “No answer”. I must to enter the command “Core restart now” in the cli before the inbound calls ring my sip phones again.

Firewall or NAT. Although as a final confirmation, you need to use tcpdump to confirm that they are really not reaching your machine.

Some ISPs also change IP addresses on established sessions, one suspects they to this to stop people running servers on home/small business accounts.

Since the morning, I have made this observation. When I edit the pjsip.conf or the extensions.conf and I go in the cli where I hit the command “pjsip reload” and “dialplan reload” the outbound calls work well not the inbound calls. But when I hit in the CLI the command “Core restart now” all work (Inbound and oubound calls) very well. Very weird… @david551 I will try the tcpdump too, for see if I have not anything else who doing that ?

The most common cause of failed incoming calls is the loss of registration.
From CLI, execute pjsip show registrations and pjsip show contacts to find out whether your trunk is still registered with ISP.

In turn, you may be losing registration due to the incorrect (too small) timers on your NAT / firewall device. Are you managing it? Can you increase UDP timeout?

Hi @Pentium-5, yes I manage this asterisk server with a colleague at my office (1-10 persons work in this company, we are a small company). But I am not a network administrator. I am graphist/web designer. But I decided to interest to the VoIP and to understand how this is work. Because, we use sip phones in the company. So, to increase the udp timeout, it is in the router or in the pjsip.conf file that I must make this modification?

it’d be in the router configuration.