What about the option of auto-answer that is available on Cisco 7960 version 5.x? Could you not setup a second line instance on the phone and set that line instance to auto answer?
Lee Goodman -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Laur Sent: Monday, August 25, 2003 11:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Intercom with Cisco SIP 796x phones? 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: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of John Todd > Sent: Monday, August 25, 2003 7:38 PM > To: [EMAIL PROTECTED] > 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" <[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 > >> > >> > >> _______________________________________________ > >> Asterisk-Users mailing list [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> -- > >> Scanned for viruses and dangerous content at > >> http://www.oneunified.net and is believed to be clean. > >> > >_______________________________________________ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
