[Asterisk-Users] This newbie gives up for now - sadly

[Christian Hecimovic]

Hi John,

I'd like to point out that there are several SIP phones that handle supervised 
transfers well. Try the Polycom phones (500 or 600). They work great.

You must remember that SIP is a fat client protocol, which is to say, the 
phones have a considerable amount of intelligence in them. When the phones 
are natively bridged (the voice streams are not traversing Asterisk), then it 
makes sense that the phones handle their own transfer and conferencing 
functionality - Asterisk is totally out of the equation. It lessens the load 
on the server. Unfortunately, it also means that SIP phones don't tend to 
interoperate super well between brands.

So, buy the "right" phones, and you'll be set.

Christian

On Tuesday 06 January 2004 05:20, John Coll wrote:
> 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
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> END OF LINE
>        -MCP
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