Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you definitely to look into SIP timers which tell how many time to resend a packet if no response is received and for how long to wait before thinking that the SIP packet got lost(network disconnected or end-point lost)
so, qualify=yes a peer means to send-keep alives and have the NAT mechanism stay active, as soon as the SIP keep-alive packets reach a no-reply (max time)x(max retries) Asterisk marks the peer as UNREACHABLE. qualify=no wouldn't do all of the above. Another interesting thing to know is that SIP end-points have registrations time-out and refresh Registration timers as well. So if everything is going well, SIP end-points refresh their registration after some defined time. On Tue, Nov 15, 2011 at 3:35 AM, eherr <[email protected]> wrote: > I think the wrap up answer is the interval of registration compacted, if > used, with the SIP OPTION packet.**** > > ** ** > > I like the SIP OPTION packet because we have scripts to monitor the status > and lets us know when a phone is up or down.**** > > ** ** > > --E**** > > ** ** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Carlos Alvarez > *Sent:* Monday, November 14, 2011 5:30 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] How do extensions "stay registered"**** > > ** ** > > I think the registration part was answered. The de-registration part is > different. If the phone is gracefully taken off line it specifically > de-registers. If it just can't be reached because it powers off or the > router closes NAT, or whatever, then Asterisk won't know this until it > times out.**** > > ** ** > > On Mon, Nov 14, 2011 at 3:19 PM, eherr <[email protected]> wrote:* > *** > > I think the question is more along the lines of how does asterisk know > immediately when a sip phone becomes on line and when you unplug the phone > from the network, how does asterisk essentially know immediately that it > status is “UNKNOWN”**** > > **** > > If I am not mistaken.**** > > **** > > --E**** > > **** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Danny Nicholas > *Sent:* Monday, November 14, 2011 5:01 PM > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] How do extensions "stay registered"**** > > **** > > “Extensions” do not register – peers do. A peer can register itself or be > registered by Asterisk. In most cases the “extension” is equivalent to the > “peer” (301 = 301) but it can be quite different (301 = sipuser1) or (301 = > [email protected]).**** > > **** > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Douglas Mortensen > *Sent:* Monday, November 14, 2011 3:52 PM > *To:* '[email protected]' > *Subject:* [asterisk-users] How do extensions "stay registered"**** > > **** > > I know this is probably a very basic question for many on this list. But > in troubleshooting an issue, I wanted to take a step back & ask the > question. In Asterisk (or maybe all SIP), how do extensions stay registered > with the SIP server?**** > > **** > > Do the extensions simply register repeatedly as a means of telling > asterisk “I’m still here”, or are there actual keepalive packets that are > transmitted to actually keep a TCP session alive? My guess is the former.* > *** > > **** > > But am I oversimplifying it? Is there more to the process?**** > > **** > > Thanks,**** > > -**** > > Doug Mortensen**** > > Network Consultant**** > > *Impala Networks Inc***** > > CCNA, MCSA, Security+, A+**** > > Linux+, Network+, Server+**** > > .**** > > www.impalanetworks.com**** > > P: (505) 327-7300**** > > F: (505) 327-7545**** > > .**** > > **** > > > -- > _____________________________________________________________________ > -- 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**** > > > > **** > > ** ** > > -- **** > > Carlos Alvarez**** > > TelEvolve**** > > 602-889-3003**** > > ** ** > > ** ** > > -- > _____________________________________________________________________ > -- 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
