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" <[EMAIL PROTECTED]> To: "Asterisk on BSD discussion" <[email protected]> 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://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- Asterisk-BSD mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-bsd _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- Asterisk-BSD mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-bsd
