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

[John Todd]

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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