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" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Intercom with Cisco SIP 796x phones?
Reply-To: [EMAIL PROTECTED]
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: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Jared Smith
 Sent: August 25, 2003 15:11
 To: [EMAIL PROTECTED]
 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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