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? Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030711/c050d65c/attachment.htm
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 <karl@putland.linux-site.net>
MGCP is more appropriate for this. It's possible that class call features could be implemented for SIP devices, but it would be even more overhead than MGCP. Mark On Sat, 12 Jul 2003, 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? > > Sean