Christian, I guess I am Confused about the 'header' stuff. I am using the SIP strictly on a LAN as extensions to the [*]. Hence, I have a line in sip.conf like this:
[2200] ;snom 200 callerid=Reception <2200> type = friend disallow=all allow=ulaw allow=alaw username = 2200 secret = 2200 host = dynamic dtmfmode = rfc2833 context=intern mailbox = 2200 In extensions.conf I have exten => 2200,1,Dial(SIP/2200,20,tT) Now, [*] is at 192.168.1.16. Where does the 'header' you refer to get sent? I tried adding intercom=true to the sip.conf but that is not it right? Lost ... Willy ----- Original Message Follows ----- > To use "Intercom" mode in the current releases of the snom > 200, you need to use an "intercom=true" flag in the > To-Header. Essentially that makes the phone to pick up the > call immediately. > > To: <sip:[EMAIL PROTECTED];intercom=true> > > However, this mechanism is likely to change because of > security concerns and new interoperable methods. > > Christian > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:asterisk-users- [EMAIL PROTECTED] On > > Behalf Of [EMAIL PROTECTED] Sent: Sunday, March 21, > > 2004 5:25 PM To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Snom 200 Voice Call / Paging > > > > To All, > > Several months (2003) ago there was a discussion > > regarding overhead paging & intercom functionality with > > SIP / Asterisk. Jerry Gibson, John Todd and various > > others participated (from checking the archives). One > > person even responded that they had the stuff working > > with the snom 200s. > > Voice Call (i.e. on-hook speaker/mic) is realy important > > in a lot of apps. It would appear that the snom 200 and > > by extension the snom 105 support the functionality. > > I will be happy to make a wiki entry to explain & demo > > this functionality once I have it working properly. I > > also understand that the (mis)use of conferencing is > > frowned upon as it wastes bandwidth and CPU. However, > > until a better way comes around, that is not a problem > > as there are quite a few applications where (a) one > > needs Voice Call (which is 1 <-> 1) and / or an > > 'allPage' which can be limited to a subset of all > > phones. Typically phones which are in designated or > > public areas, conference rooms, etc. The BW/CPU issue > can be controlled. Better a limited solution than no > > solution at all ;) > > I am also allowing for the limitation that all > > participating phones are on the same LAN with the [*]. > > Anyone who has this successfully working with snom, > > please respond .. Using the [*] sound card for a > > separate PA system is NOT an option ;) > > As I said, I will be 'distilling' the info and turn it > > into a wiki entry. > > Cheers and TIA, > > Willy > > > > Willy Wouters > > ypOne Publishing > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
