No subject

will be in the Asterisk call control integration.

All this hinges on the fact that all the XML functionality built into
the CallManager phone load is also built into the recent SIP phone
loads.  I guess trial and error is the best way to find this out.

Good Luck!

Ray Burkholder
One Unified
519 570 0689 x2002


> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com=20
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of=20
> Jared Smith
> Sent: August 25, 2003 15:11
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] Is Asterisk ready for "real" use?
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>=20
> Oh really?!?  Can you give us more information...
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> On Mon, 2003-08-25 at 12:30, Ray Burkholder wrote:
> > The Cisco SIP phones have a second voice channel available=20
> for a paging
> > type of implementation.  Now the problem is simply of=20
> finding someone
> > and some time to see if it can be made to work with Asterisk.
> >=20
> > Ray Burkholder
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