R: R: R: [Asterisk-Users] AT-320 + supervised transfer

[Giordano Grandis]

Hi Gawin,
I'm newly testing the atxfer and i always the same question: if i transfer a call to a peer that don't answer me, ho can i re-take the call.
Actually i got the call hanged up without the possibility the speack back with my first caller.

Thanks
Giordano                    
                     

-----Messaggio originale-----
Da: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Per conto di Gavin Hamill
Inviato: mercoledì 1 giugno 2005 15.31
A: asterisk-users at lists.digium.com
Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer

On Wednesday 01 June 2005 14:15, Giordano Grandis wrote:
> This is what happen when i call a peer that not answer:

> Jun  1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: 
> Unable to create channel Local/406 at local/n do you have chan_local?

I don't like the look of this part at all. Please try to "rm /usr/lib/asterisk/modules/*" then 'make clean; make install' on a fresh checkout of CVS HEAD :)

Also, there should be no need for the 'r' option to Dial since SIP already supports all the progress indication necessary.

gdh
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