Robert Hajime Lanning: "He is using SIP phones. Supervised Transfers do not really work with SIP. He wants, on a SIP phone (I think he had Grandstream phones), to: o hit "transfer" o dial new extension o talk to new extension ***** this part does not work ***** o hit "transfer" to complete the transfer or some cancel button to abort"
Yes that is exactly what I want - thanks for clarifying. -------------------------------------- Derek Irwin: "I guess what I'm saying is from the start, * continues to surprise and impress. If you put in the time to learn it, you will be rewarded with a feature-rich system that can go head-to-head with the commercials system out there." Well perhaps Derek but my experience so far, and I'm not talking rocket-science requests, is that Asterisk just does not do the most basic of things "out of the box" and that the "documentation" is so dispersed and incomplete that it needs a massive effort to get even the most basic stuff running. And in some cases even the most basic stuff turns out not to work - yet. I will come back to asterisk when it is "leading edge" and not "bleading edge". This is not a criticism of asterisk - just that its clearly not at the stage where an average linux sysadmin can use it for normal PBX applications with a reasonable time investment - if at all. I am sure it will get there and I am very keen to come back on board when it does. I hope I have not offended any developers by these comments - I know I am just sitting here while you guys do all the work. Please keep up the good work - and thanks for the comments. john <quote who="Tilghman Lesher"> > On Monday 05 January 2004 13:44, John Coll wrote: >> This newbie has been trying out Asterisk. It has been both a) >> surprisingly painful and b) impressive in terms of helpful support >> from other users. >> >> Having got two phones to communicate and then got voicemail MWI >> going (neither painlessly) I decided the next step was to implement >> call transfer as per nearly all commercial PBX systems i.e. >> >> hold call >> consult another extension >> either exit and let the two speak >> or get back the original caller >> >> - an utterly fundamental office procedure on a PBX. > > I don't know why you'd need to implement that, as it's as simple as > turning on two options in zapata.conf. Actually, I think both of > those options are on by default in the sample configuration files. > >> And I've spent the requisite few hours on Google and all the docs I >> have printed out. Eventually I found the thread "transfer with >> three-way calling" (circa Mon, 15 Dec 2003 20:45:08 -0600) and it >> seems that I can't do that basic operation in Asterisk. > > Why not? Are you not able to send a flash hook? > > -Tilghman > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > -- END OF LINE -MCP _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
