[Asterisk-Users] Intercom with Cisco SIP 796x phones?

[John Laur]

BTXML support for client applications is necessary to achieve this. The
SIP images state that they support BTXML; however, they only use it for
their internal screens and internal navigation. CMXML is the only
language supported for client applications with the SIP loads currently.
A little bird at Cisco told me that a future version of the SIP loads
will support BTXML client applications. (and a less reliable bird told
me that the next version of call mangler will be SIP-based)

This will support all of the good stuff I really want to be able to do
with this phone. For instance, we could forward a URL with the call to a
BTXML app that causes the phone to display extended information about
the caller, or the voicemail Services button could show the users a menu
of their voicemails and choosing one would play the message directly
over the speaker. Also, intercom would be supported. All this is
documented in the BTXML guide and ready to go whenever they "open it
up"..

I suppose it might also be possible to find an "exploit" in the current
firmware that causes the phone to execute some BTXML from a remote
location.. we might be able to get in that way, too :)

John

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of John Todd
> Sent: Monday, August 25, 2003 7:38 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Intercom with Cisco SIP 796x phones?
> 
> If you find a way to make the phone request that second audio stream
> without user intervention, I'm all ears.  :-)
> 
> JT
> 
> 
> At 5:15 PM -0400 8/25/03, Ray Burkholder wrote:
> >From: "Ray Burkholder" <ray at oneunified.net>
> >To: <asterisk-users at lists.digium.com>
> >Subject: [Asterisk-Users] Intercom with Cisco SIP 796x phones?
> >Reply-To: asterisk-users at lists.digium.com
> >Date: Mon, 25 Aug 2003 17:15:01 -0400
> >
> >I read about this intercom stuff on page 62 & 63 of the book
"Developing
> >Cisco IP Phone Services" isbn 1-58705-060-9.  Primary calls take
place
> >on streaming channel 0.  When streaming channel 0 is not in use,
> >streaming channel 1 can be used for asynchronously streaming (in and
> >out) stuff like voicemail, email, and, yep the one we want, intercom.
> >Page 87-88 of the book talks about CiscoIPPhoneExecute to push the
> >commands to the phone.
> >
> >On the last two pages of an addendum found at
> >http://services.dogma.net/errata.doc, more details are provided for
> >connecting to streaming port 1.
> >
> >http://cisco.evolvis.net/ivision/pdfs/Jukka_Nurmi_iVision2003.pdf
> >provide some background on Cisco's IP Phone Services.  Title is
foreign
> >language, but text is English.
> >
>
>http://www.loligo.com/asterisk/Cisco/79xx/2003-06-20.from-ftpeng.cisco.
c
> >om/CMXML_App_Guide.pdf provides additional program details.
> >
> >>From what I see, basic functionality should be a piece of cake.  The
fun
> >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
> >>  [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> >>  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?
> >>
> >>
> >>  Oh really?!?  Can you give us more information...
> >>
> >>  On Mon, 2003-08-25 at 12:30, Ray Burkholder wrote:
> >>  > The Cisco SIP phones have a second voice channel available
> >>  for a paging
> >>  > type of implementation.  Now the problem is simply of
> >>  finding someone
> >>  > and some time to see if it can be made to work with Asterisk.
> >>  >
> >>  > Ray Burkholder
> >>
> >>
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