I hate to bring up an old thread, however, I'm implementing SLA as well. I've got SLA working, my tunk executes slatrunk(line1), and my polycom 650 phone rings on the SIP subscribed line (button 1)
I'm assuming slatrunk sends the calls to the SIP/station1 SIP device, so the call will always appear on "button 1" on the end device? My question then is how does the receptionist or answering party know which line this came in on so they can, for example, overhead page and say "Fred, you have a call on line 1?" My next observation is on the execution of slatrunk() the call quits ringing, so the calling party gets dead air. Is that normal? I believe the call coming in on button 1 is hear to stay, part of the 'trickery' but if their is an easy way to know which line that call is from that would help, and secondly if slatrunk played a ringing tone till it saw a station join the bridge that could help. And perhaps I have everything misconfigured as well :) Thank you for all the asterisky things you do! Brian On Fri, Nov 30, 2007 at 6:10 PM, Russell Bryant <[EMAIL PROTECTED]> wrote: > Mark Wiater wrote: > > I fought with this in 1.4.5 with polycom phones. I was hoping to share a > DID from a PRI on several > > Polycom IP430's. > > > > Might you be willing to share some specific configurations for such a > situation? > > There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I > have > on my to-do list to spend a week with an SLA test environment and coming up > with > an extensive set of examples of the different ways it can be used. > > I will post something to this list when that is available. > > -- > Russell Bryant > Senior Software Engineer > Open Source Team Lead > Digium, Inc. > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- Brian McManus
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