[asterisk-users] How do extensions "stay registered"
Sammy Govind
govoiper at gmail.com
Mon Nov 14 23:35:43 CST 2011
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.eherr9633 at gmail.com> 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:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *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.eherr9633 at gmail.com> 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:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *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 =
> doug at impalanetworks.com).****
>
> ****
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Douglas Mortensen
> *Sent:* Monday, November 14, 2011 3:52 PM
> *To:* 'asterisk-users at lists.digium.com'
> *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****
>
> .****
>
> ****
>
>
> --
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>
> ****
>
> ** **
>
> -- ****
>
> Carlos Alvarez****
>
> TelEvolve****
>
> 602-889-3003****
>
> ** **
>
> ** **
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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