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
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:123@bla.com;intercom=true> However, this mechanism is likely to change because of security concerns and new interoperable methods. Christian> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of willy@yponeinc.com > Sent: Sunday, March 21, 2004 5:25 PM > To: asterisk-users@lists.digium.com > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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:123@bla.com;intercom=true> > > However, this mechanism is likely to change because of > security concerns and new interoperable methods. > > Christian > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users- admin@lists.digium.com] On > > Behalf Of willy@yponeinc.com Sent: Sunday, March 21, > > 2004 5:25 PM To: asterisk-users@lists.digium.com > > 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 > > Asterisk-Users@lists.digium.com > > 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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersWilly Wouters ypOne Publishing