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

[John Coll]

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
>
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>


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