I don't quite understand the use case, but it sounds like you may be trying to do shared line appearances (http://asterisk.org/node/48342). You seem to be alluding that you want multiple extensions to share the state of a single extension. If that is the case, then SLA isn't quite that. Also, Asterisk SLA doesn't support a notion of call appearance where a single extension can receive multiple calls.
-- Raj On Mon, Mar 10, 2008 at 11:00 AM, Tony Plack <[EMAIL PROTECTED]> wrote: > I am working on a project that requires shared extension. Where shared line > looks at the status of a line/trunk, shared extension would look at a series > of channels as the same "extension". > > The users would like to add destination channels on the fly, to provide > roaming extensions, but maintaining fixed channels as well. > > If a call comes in on an extension, the system needs to honor the fact that > channel 1 is busy, therefore, the extension is busy. Keep in mind that the > channel could be anything including SIP outbound trunk channels (read cell > phone or hotel room). > > The Dial command does provide a nice multi-channel dialer, especially with > the "r" option, however, if one of the lines is busy, the system will keep > ringing the other lines until timeout or answer (read voice mail). > > So I am contemplating adding a feature to the dial command, that would make > any channel busy, cause the initial Dial to come back as busy. Kind of a > force the state flag. > > Before I brake into code, does anyone have any other ideas? > > This would also help with phones like Grandstream, where you have 4 accounts > to configure, and would like to have all 4 SIP accounts act as 1 extension. > > Tony Plack > > _______________________________________________ > -- 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 > -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org _______________________________________________ -- 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
