> Is there any way to force this? I have several user agents and I want to > achieve > near 100% availability for all peers. I realise that the peer will be 'woken' > up > at my qualify intervals, but can I actually force registration from the CLI?
For those peers which are at known, fixed, predictable IP addresses (e.g. in-house phones which have statically-configured IP addresses or which get non-dynamic addresses from a DHCP server you control) you do not need to use registration at all. You can simply hard-code the peer's address into sip.conf (or, I presume, an equivalent realtime table). When you Dial() such a peer, Asterisk will start sending out the INVITE packets, regardless of whether it has heard anything at all from that peer in the last hour or fifty. No need for "qualify" although you can use this to keep track of whether the peers are actually alive or not. If you take this approach, you'll save yourself a great deal of heartburn if you can figure out an automated way of keeping the IP addresses synchronized, between Asterisk and whatever "hand out the addresses" mechanism the phones use (DHCP, TFTP-based provisioning files, etc.). Keep a "master list" of peers and addresses in a simple table or file somewhere, and use this to populate the other pieces of software which need to know. For peers which can move around to arbitrary IP addresses, and where your server system won't know what those addresses may be in advance, using REGISTER from the device is really the only good approach. If you've got a setup where devices change their IP address frequently and need to be on-line constantly, I'd say you have a fundamental problem with no easy solution. Using a short registration time limit (e.g. 30 seconds) is probably the least awful way to handle this, and if you're talking about a very large number of phones you may want to set up a dedicated SIP proxy to handle this registration burden and keep Asterisk from having to deal with it. -- _____________________________________________________________________ -- 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