Ray Burkholder
2003-Aug-25 14:15 UTC
[Asterisk-Users] Intercom with Cisco SIP 796x phones?
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.
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@oneunified.net> >To: <asterisk-users@lists.digium.com> >Subject: [Asterisk-Users] Intercom with Cisco SIP 796x phones? >Reply-To: asterisk-users@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@lists.digium.com >> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of >> Jared Smith >> Sent: August 25, 2003 15:11 >> To: asterisk-users@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 >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> 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 >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users