[asterisk-users] call transfer not working

[Rizwan Hisham]

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
>
>
> ------------------------------
> Get the Yahoo! toolbar and be alerted to new email
> <http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.php>wherever
> you're surfing.
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best Regards
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070704/400c55b2/attachment.htm