Inbound calls stop working after 60 seconds

Hello,

I am running version 2.8, my remote SIP devices are loosing connections (well sort of). After my remote SIP device boots, I can make inbound/outbound calls, but after about 60 seconds I can only make outbound calls. When I make inbound call to the SIP extension, I get “You have reached the number that has been disconnected…”.

This is the message I get after about 60 seconds or so:

[Jul 15 16:54:51] NOTICE[3156] chan_sip.c: Peer ‘674’ is now UNREACHABLE! Last qualify: 112

And this is what I get in the trace when I make inbound call:

[Jul 15 16:55:15] WARNING[6730] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Jul 15 16:55:15] VERBOSE[6730] logger.c: == Everyone is busy/congested at this time (1:0/0/1)

Again, even after all this I can still make outbound calls from the SIP device, only the inbound calls to the SIP device do not work. I am new to Asterisk, and to me it looks like as if the remote SIP device ‘Keep-alives’ are timing out. I am able to produce this behavior on a softphone and a polycom IP phone.

I would appreciate any help/suggestions I can get on this.
Thanks.

Hi,

I am still having the same problem, can anyone please point me in the right direction or make some suggestions?

Thanks.

Asked and answered numerous times here on the forum.

Check your NAT settings and that your firewall is open for UDP 5060 and UDP 10000 - 20000

Mikael,

Thanks for your response. The ports are open as you suggested, in fact I opened any/any to my server to rule out any firewall issues. Whats strange is that, the outbound calls from my remote SIP devices always work, its only the inbound (calls to remote SIP devices) that stop working after about 60 seconds or so. Is there a config file where I can tweak the SIP keep-alive settings?

You have a NAT issue. Please read these articles:
http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
http://www.voip-info.org/wiki/view/Asterisk+sip+nat