We do not send a reinvite when you put a sip call on hold.
if you have the hold-music param set in the settings section
or in each span you can deliver moh to the on-hold party.
The call swapping concept and the 3-way feature are
competing features so we would need to design config
options to expose them.
Judging from your numerous feature requests on TDM
you should consider sponsoring the OpenZAP project because
our resources are limited and TDM users such as yourself
should appreciate the goals of a BSD-licensed unified TDM
library. Everything you see so far in OpenZAP has been
done out of faith that sponsors will step up and try
to steer the project towards it's eventual goals of
bigger and better ISDN support/ss7
etc.
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
----- Original Message ----
From: sanjeev mudholkar <sanjeev.mudholkar@gmail.com>
To: freeswitch-dev@lists.freeswitch.org
Sent: Friday, January 25, 2008
1:18:29 AM
Subject: [Freeswitch-dev] Call Hold ans swapping on FXS
Hi,
I'am trying to use Freeswitch(with FXS) as a UA. Everything is
fine for a outgoing and incoming call.
1)On doing a hold (flash) on the FXS the call is put on hold
but there no SIP signaling (i.e re-invite is not sent to the peer).
How can I force freeswitch to send a re-invite?
2) I'am not able to establish a consultation call as it is conferenced with the first
one as soon as the second call is connected. This also prevents
from swaping between calls. The call swapping feature seems to have
been removed by introducing the transfer/conference feature.
Is there away to achieve call swapping ?
Appreciate your help.
Thanks in advance
-Sanjeev