On Fri, 2003-07-11 at 22:12, Sean P. Robertson wrote: > > Here is a question that needs a few opinions... > > Recently we installed a couple of FXS gateways into a site with a SIP > Proxy/Softswitch other than Asterisk. One of the things that the > users on this site need to do is receive calls on single line phones > on the FXS gateways and then hookflash and transfer them to other SIP > users. > > We found that the FXS units, true to their nature as VoIP gateways, > saw the hookflash and passed a SIP INFO (event hookflash) back to the > Proxy. The Proxy sent this message on to the calling SIP phone which > replied that this "feature is not implemented." > > The gateway manufacturer says that the Proxy should process the INFO > packet, place the calling endpoint on hold (as a PBX would), stream > dialtone to the gateway prompting the user to dial the digits > indicating the destination to whom the calling party should be > transferred, and then do a transfer. > > The Proxy manufacturer says that the gateway should see the > hookflash, Hold the caller locally (as a SIP phone would), and give > new dialtone to the single line phone prompting the user to dial the > digits digits indicating the destination to whom the calling party > should be transferred, and then send a complete transfer sequence to > the Proxy. > > My question is, how would Asterisk handle a situation like this? Are > there any opinions as to how this scenario should be handled?
Asterisk currently only handles dtmf INFO messages. --Karl > > Sean -- Karl Putland <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
