--- picciuX wrote: > maybe you can try to issue a "sip show registry" on > the console on a regular > basis and watch if your * loose registration.
Ok: asterisk1*CLI> sip show registry Host Username Refresh State proxy01.sipphone.com:5060 17476510045 105 Registered Also, at my.sipphone.com, when I log in and view advanced features, in "SIP Registrations" the status is always "on line" and "Public IP address" shows the IP address of the NAT device which my asterisk machine is behind, followed by e.g. "(expires in 1020 seconds)". According to both asterisk and sipphone, I'm never losing registration. > You can also turn on sip debug on the console, to > see if the "unanswered > calls" effectively reach asterisk or not. I did "sip debug" on the console and got "SIP Debugging enabled". Now, every 20 seconds or so, I get: <-- SIP read from 192.168.3.22:5060: --- (0 headers 0 lines) Nat keepalive --- Trying to call after enabling debugging, some calls succeed and some fail (as usual), and I get no indication of the call attempts on the console when the call fails. Every minute or so, I get long spiels on the console (unrelated to the timing of my call attempts) starting with: REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 198.65.166.131:5060: REGISTER sip:proxy01.sipphone.com SIP/2.0 which sometimes contain strange things like: Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> <-- SIP read from 192.168.3.22:5060: --- (0 headers 0 lines) Nat keepalive --- asterisk1*CLI> <-- SIP read from 198.65.166.131:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.23:5060;branch=z5lR3bK653e4bb;rport=1937;received =72.171.58.49 From: <sip:[EMAIL PROTECTED]>;tag=as741dda96 To: <sip:[EMAIL PROTECTED]>;tag=21a68532c2cd5d9b34affe6bba40a2e. 1bb5 Call-ID: [EMAIL PROTECTED] CSeq: 166 REGISTER P-Behind-NAT: Yes Contact: <sip:[EMAIL PROTECTED]:1936>;q=0.00;expires=26 Contact: <sip:[EMAIL PROTECTED]:1937>;q=0.00;expires=120 Content-Length: 0 --- (10 headers 0 lines)--- Scheduling destruction of call '[EMAIL PROTECTED] ne.com' in 32000 ms Which is strange because I had no incoming or outgoing calls or call attempts at the time I got those messages on the console, yet asterisk is talking about destroying calls. > In the > latter, is sipphone that > loose your registration, Yes, this appears to be the case. > so you maybe can lower the > time before registration > renewals. But during the time I was doing tests and recording the above information to put in this message, I had several call attempts succeed and several fail, and several minutes later, the SIP registration I mentioned at my.sipphone.com was down from 1020 seconds to 681 seconds, and then later I checked again and it was down to 412 seconds, etc. So all the while when I was having some calls succeed and some fail, my sipphone registration had not yet been renewed (according to sipphone). So I don't know what all the registration stuff is that asterisk is dumping to the console in debug mode, but it's apparently not reregistration with sipphone, since sipphone's timer doesn't get reset by it, and it doesn't seem to have any relationship to whether my incoming call attempts succeed or fail. > And turn on "qualify=yes" for your peer to > keep fresh nat mappings > on the router. I tried that yesterday, and it seemed to have no effect. Based on this information, can you give any clue as to what the problem might be? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
