From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify "If you turn on *qualify* in the configuration of a SIP device in sip.conf<http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf>, Asterisk will send a SIP OPTIONS<http://www.voip-info.org/wiki/view/SIP+method+options>command regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER function<http://www.voip-info.org/wiki/view/Asterisk+func+sippeer>, and inversely this function will only provide status information for peers which have *qualify=yes*." My guess is that your Nat/firewall is closing the connection after some time the phone is idle, so this way Asterisk will make sure to always have communication going trhough that connection so your NAT/firewall won't just close it.
try playing with qualifyfreq as well. Let us know if it helped. Alyed 2010/3/27 James Lamanna <[email protected]> > Hi, > I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. > After some period of time, asterisk says that some of them are > unreachable, and the phones lose their registration. > The only way to make the phones recover is to clear the NAT > translation tables for the phones on the PIX (clear xlate...) > Does anyone know how to fix this? As you can imagine, it is quite > annoying. And it does not happen to all the phones either. > > sip fixup is enabled on the PIX > > phone config parts: > > nat_enable : 1 > nat_received_processing : 0 > nat_address: [public ip of PIX] > > Thank you. > > -- James > (Please CC me on all replies) > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
