I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a number of SIP clients on a LAN being natted. If I open a single client on the LAN, it all works as expected. However, if another machine on the LAN opens a client no client will work. Attempting to call anything like Voicemail fails and after a short while Asterisk starts scrolling: [Feb 10 11:10:31] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-5c0 at 192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Feb 10 11:10:31] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-5c0 at 192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. [Feb 10 11:10:32] WARNING[8852]: chan_sip.c:3779 retrans_pkt: Maximum retries exceeded on transmission 1064dc5c-5101a8c0-13c4-3ba4-e88578-5c0 at 192.168.1.81 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. The only way to get service back is kill all other clients on the LAN and restart the router. Naturally I'm questioning the router, but the fly in the ointment is that it worked before I upgraded from 1.6.1 to 1.6.2 - which makes me think that it could be Asterisk itself. I'm starting to wonder if there is an issue in the Asterisk NAT code as I'm also seeing some 'stale nonce received' relating to the LAN IP of the second client after I disconnect it. I'm struggling to work out how can I debug this effectively and would appreciate some guidance here.
Warren Selby
2010-Feb-10 19:10 UTC
[asterisk-users] Nat Issue - is this Draytek || Asterisk?
On Wed, Feb 10, 2010 at 5:53 AM, Brian < brel.astersik100129 at copperproductions.co.uk> wrote:> I'm struggling to work out how can I debug this effectively and would > appreciate some guidance here. > >Try enabling sip debug on the internal peers (sip set debug peer xxxx from the cli) before you bring the second peer up, and go from there? -- Thanks, --Warren Selby http://www.selbytech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100210/ba46c1c0/attachment.htm