[Asterisk-Users] SIP Attended/Supervised transfer & features.conf

[C F]

Make sure you put t (for the called party) or T (for the calling
party) in the Dial command option, like this
Dial(somephone,timeout,tT)

On 4/11/05, Gonchi Mateos <g-mateos at adinet.com.uy> wrote:
> Hi all,
> 
> We were willing to try the SIP Attended/Supervised transfer with * realease
> 1.0-7. From the wiki´s feature.conf config page we found that a special
> section called featuremap had to be added to the config:
> 
> [featuremap]
>  blindxfer => #1                ; Blind transfer
>  disconnect => *0               ; Disconnect
>  automon => *1                  ; One Touch Record
>  atxfer => *2                   ; Attended transfer
> 
> We made that changes but upon pressing *2 nothing happens, neither with
> #1 for the blind transfer. The blind transfer is working as it defaults
> in *, with # plus extension.
> 
> We tried to unload and reload res_features module but with no luck as it
> says that the user count is 1.
> 
> After some examination at chan_sip.c, we found the supervised transfer code
> section, but we found nothing on the parsing of the featuremap section.
> We did find the parsing of the first section of the config file concerning
> call parking, which does work.
> 
> Any idea on how to make it work?
> 
> Thanks to all,
> Gonchi
> 
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