[asterisk-users] problem transferring calls some of the times

[Ian]

Hi Raul

I have bypassed my Grandstream's transfer function, by enabling "*2" 
transfers in features.conf, and setting "canreinvite=no" in sip.conf

Hope this helps you

Ian

Raúl Gómez C. said the following on 03-Mar-08 08:34 PM:
>
>     In the config file (sample) "features.conf" are some commented
>     lines that said:
>
>     /"; Note that the DTMF features listed below *only work when two
>     channels have answered and are bridged together*.
>     ; They *can not be used while the remote party is ringing or in
>     progress*. If you require this feature you can use
>     ; chan_local in combination with Answer to accomplish it."/
>
>
>
> BTW: I don't have a clue how "/can I use chan_local in combination 
> with Answer to accomplish it."/, so if anyone knows please give some 
> help!
>
> Thanks in advance...
>
> -- 
> Raul
> Linux Counter #156439
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