Woof wrote: > On Mon, 30 Nov 2009 10:01:39 -0500, Paul Mossman > <[email protected]> > wrote: > > > Woof commented in XX-7151 that: Configuration of per-user > MOH can be > > moved to a separate MOH configuration file or directory, > and need not > > effect alias.xml at all. (Thanks!) Just to confirm > though, changes > > can take effect without a service restart? > > Depends how the information is conveyed to sipXivr. Today it > checks the modification date on most of it's configuration files (like > validusers.xml) and reloads and caches them on the next call > after they change. No restart required.
Avoiding service restarts would be key. (Users are able to invoke these changes.) If we *must* restart something, it should be the Phone. > > That brings me to another proposal... I think "None" > should be an MOH > > Audio Source option. This wouldn't change the MOH URI. > What it would > > do is result in the held party not receiving any audio when held. > > (Whatever the acceptable way to accomplish this with the SDP is...) > > That one would require phone config changes, so it doesn't > invite the MOH server in the first place...or perhaps the MOH > server could reject the MOH call. However, to accomplish the > same thing, silence could be played as an MOH source for > those situations. It's not the same from SDP and RTP point > of view, but the caller wouldn't notice the difference and > the effect would be the same. The phones would still invoke MOH as usual, since they're configured with a MOH URI. But yes, make sure the held party hears only silence. > > It would be very nice for a user to have the ability to disable MOH > > after dialing into a conference call, without having to > disconnect and > > dial back in. > > > > (Bonus points if MOH source configuration changes can be applied to > > calls already on hold.) > > Not sure how you'd signal that. OK, applying changes to calls already on hold is getting carried away. :) But I didn't mean to imply conference-specific behaviour. That was just an example. Here's how I see it. The phone always have the same (user-specific) MOH URI, so they don't need to be rebooted when the Audio Source configuration changes. It's the MOH server(s) that apply different behaviour based on the configuration. Upon establishing the dialog for an INVITE to ~~mh~juser, the Audio Source currently configured for 'juser' is used. The MOH server doesn't care if the receiving endpoint is a conference, gateway, phone, etc. -Paul [email protected] _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev sipXecs IP PBX -- http://www.sipfoundry.org/
