check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general
settings.
On 7/4/07, satish patel <satish_patel_2000_2000 at yahoo.com>
wrote:>
> Dear all
>
> I have install asterisk 1.2.x and it is working fine my
> setup is like
>
> [*]-------[Mediant2k]------------[Avaya]
>
> Now i want to transfer call in internal extension i have read more
> document on www.voip-info.com but it is now so much clear so if u have any
> sample configuration file and doucment plz suggest me i have configure
> feature.conf and extention.conf for this task
>
> regards
>
>
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Best Regards
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com
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