[Asterisk-bsd] Pickup reinvite

[Tony Jago]

Welcome to the wonderful world of nat. "No response to our critical packet"
is telling you there is a problem with your SIP signaling (not the RTP
stream). It means that asterisk is sending out a SIP packet, probably the
re-invite and getting no response back from the other device. This is
probably becase the NAT is blocking the packet. Every different NAT router
handles NAT and SIP differently. You probably need to setup port forwarding
for each of your sip device behind the nat. This means changing the SIP port
and the RTP port range for every phone. The phone need to be an static IP's
and the ports forwarded through the router.

The other option which *may* work is to have your phones keep the nat
session alive by constantly sending a SIP packet to the asterisk server.
This is a bit of a hack and isn't as reliable as setting everything up
correctly with port forwarding.

Tony

----- Original Message ----- 
From: "Tim St. Pierre" <tim at communicatefreely.net>
To: "Asterisk on BSD discussion" <asterisk-bsd at lists.digium.com>
Sent: Tuesday, December 11, 2007 8:40 AM
Subject: [Asterisk-bsd] Pickup reinvite


Hello Folks.

I'm wondering if anyone has any helpful hints.

I recently upgraded to 1.4.11, and I'm having problems with pickup, both
directed, and the pickup feature.

My server is on the public internet, and all phones are behind a NAT router,
somewhere else on the public internet.

When a ringing phone is picked up by another phone, you have audio for a few
seconds, then the call is dropped.

The console shows "No response to our critical packet"

A SIP debug of the conversation between the phone and the server shows a
re-invite request right when the call drops.  The phone is of course using
the internal IP address as it's contact, and it looks to me like the server
is trying to use it.

I have canreinvite=no for both the general sip.conf, as well as per-peer.

I am using the whole range of Aastra Enterprise IP phones.

Interestingly enough, some phones show their true IP address and port in the
Asterisk registration database.  I believe this is where the phones have
successfully communicated with a uPNP router, and discovered their public
address.  These phones can successfully pickup the call.

If I pipe the pickup call through the Local channel, it works.

Why is asterisk still trying to re-invite even though I have explicitly told
it not to in the config?

It worked fine in 1.2

Any suggestions, or requests for more information?

Thanks for any help.

-Tim
-- 
Tim St. Pierre

IP telephony specialist
sip://5101@communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim at communicatefreely.net

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