check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general
settings.
On 7/4/07, satish patel <[EMAIL PROTECTED]> 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
_______________________________________________
--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
