SIP2SIP

Thanks, yes, it's working with your command. So you have to specifically state sip and can't just use ANY, ok.

Asterisk was having a problem where it wasn't going to sipthor at all, acted like it didn't know it was even there and registration kept failing. So I made a bunch of changes in the code as explained in the post.

This started about 2 days ago so I am not sure if you made any changes then. If it's not the SRV record problem then I don't know what it was but it would not send packets to sipthor.

Is it possible that there was a change that makes Asterisk not pick up the SRV info anymore?

Also, please look at the things I said about the info you give for Asterisk configuration, those settings make it send you a lot more OPTIONS packets than really needed and takes more of your bandwidth (not that it's a problem on the user side of course but your server suffers).

It's possible that the OPTIONS problem is because Asterisk should be sending those to sipthor but I had to reconfigure it so it may still be a little screwed up.

Thanks.

Two days ago we removed the SRV _sips._tcp.sip2sip.info since we disabled TLS due to instability. In the NAPTR you see it has the highest number, so the other two SRVs should be picked before the 40 100 "s" "SIPS+D2T" "" _sips._tcp.sip2sip.info.

If you have a better working asterisk config we are happy to share it on the wiki.

Thanks. That may have been what triggered this problem. I will have to go look in the source code and see how it pulls out the SRV record.

It should be only asking for the udp one because the SIP code doesn't do TCP yet.

As for your configuration on the wiki, mine was originally working fine with just one [sip2sip-0] entry like you have there and NO "outboundproxy=proxy.sipthor.net".

The other entries [sip2sip-1] [sip2sip-2] if added cause OPTIONS packets to be sent to all your proxy servers, leaving the NAT routes active, which may be required in some cases, I can't say, it was working perfectly for a while before this happened. But I think if it just had one extra entry by IP address [sip2sip-1] that should be enough to keep that NAT route open to one proxy.

The question I still have is, should the sip2sip.info server get REGISTRATION packets and OPTIONS packets sent to it (with response) in a normal situation?

Yes register packets should also arrive at proxy.sipthor.net (which are actually three IPs) (You'll get proxy.sipthor.net form sip domain sip2sip.info by doing NAPTR lookup, SRV lookup, A lookup)

Thanks. It looks to me that there is no server at sip2sip.info to answer, which should be correct since any sip traffic should go to proxy.sipthor.net

I also found that Asterisk is looking up (or attempting) _sip._udp.sip2sip.info in the source code.

Sorry for the problems, I am going to make some code changes and try to submit them to the Asterisk guys and hope they get in there.

I consider this resolved so you can close this issue, thanks.