[Asterisk-Users] three way call using sip

C F shmaltz at gmail.com
Fri Jan 21 13:01:05 MST 2005


You are right that according to theri web site it does, however it
just doesn't work.
The following is an email I received from them:
>From me:
Thanks for your reply, I figured it out. However I have another
problem, is the Conference button suppose to work? Also when putting
someone on hold if I hang up the call gets disconnected, it should
either stay on, or the hold button should be programmable.
Thanks again.
This is their response:
Hi,
 
The Conference button does not work and the Hold button is not
programmable at this time.
 
Thanks,
 
Cindy Li
Technical Support
Grandstream Networks, Inc. 




On Fri, 21 Jan 2005 13:34:13 -0600, Eric Wieling <eric at fnords.org> wrote:
> Paul Rodan wrote:
> 
> > The BT100's do support conferencing, most SIP phones do. But how does your
> > Asterisk connect you to the PSTN? Through a Zap interface? If so, what kind;
> > or through a VoIP provider like BroadVoice, NuFone, LookieLoo, VoipJet,
> > VoicePulse?
> >
> > You basically need to make sure your Asterisk server has access to more than
> > 1 line. If it does, then you should be able to 3-way call without any
> > problems.
> 
> According to this only the BT102D supports "conferencing" aka 3-way
> calling:
> 
> http://www.grandstream.com/Product_Spec.pdf
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



More information about the asterisk-users mailing list