Some incoming calls lost


I set up asterisk, with 2 sip providers.
With the first provider everything runs just fine, but with the second one, some of the incoming calls don’t reach my asterisk. (except that it runs fine too)

I first tought it was a problem with that provider but when I set it up with a softphone (x-lite) directly, it works fine.

the 2 providers have exactly the same context.

any clue ?

I guess you mean i didn’t provide enough information ?

I’m using Asterisk 1.4.1
I’m behind a NAT

When the calls don’t come trough I have absolutely nothing in the log (neither see no packets with tcpdump)

I could post snippets of my conf files but i’m not sure witch ones would be relevant.

turn on sip debug for the peer and see what you get in the way of sip packets. if nothing, i would suspect your port forwarding rules either aren’t in place, or your NAT is not keeping a hole open (are you using a qualify statement ?) or there is something else blocking the traffic.

Thank you for your answer.

When the calls don’t come trough I have nothing in the sip debug.
My port forwarding rules are set up (5060, 5061 and the rtp port range)

I’m using qualify=yes

My guess was also the hole in the NAT wasn’t kept open. But why does it work fine for the other provider, and why do some calls come trough ?

Or could it not be that Asterisk doesn’t properly register with the provider and at some moments isn’t registered anymore ?

I’m having the same problem.

I investigated and asterisk lacks of a really important feature, a way to define the registration expiration for EACH sip provider.

if you set defaultexpiry to please one of the sip provider for example (every 30 mins) the other who expect it every 5 mins will assume you’re not connected anymore.

too bad that defaultexpiry only works in general context.