<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:courier,monaco,monospace,sans-serif;font-size:12pt">We do not send a reinvite when you put a sip call on hold.<br>if you have the hold-music param set in the settings section<br>or in each span you can deliver moh to the on-hold party.<br><br>The call swapping concept and the 3-way feature are <br>competing features so we would need to design config<br>options to expose them.<br><br>Judging from your numerous feature requests on TDM<br>you should consider sponsoring the OpenZAP project because<br>our resources are limited and TDM users such as yourself<br>should appreciate the goals of a BSD-licensed unified TDM<br>library. Everything you see so far in OpenZAP has been<br>done out of faith that sponsors will step up and try<br>to steer the project towards it's eventual goals of<br>bigger and better ISDN support/ss7
etc.<br><br><br><br><br><div> </div><div>Anthony Minessale II<br><br><span>FreeSWITCH <a target="_blank" href="http://www.freeswitch.org/">http://www.freeswitch.org/</a></span><br><span>ClueCon <a target="_blank" href="http://www.cluecon.com/">http://www.cluecon.com/</a></span><br><br>AIM: anthm<br>MSN:anthony_minessale@hotmail.com<br>GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>IRC: irc.freenode.net #freeswitch</div><div><br>FreeSWITCH Developer Conference<br>sip:888@conference.freeswitch.org<br>iax:guest@conference.freeswitch.org/888<br>googletalk:conf+888@conference.freeswitch.org<br>pstn:213-799-1400</div><div style="font-family: courier,monaco,monospace,sans-serif; font-size: 12pt;"><br><br><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;">----- Original Message ----<br>From: sanjeev mudholkar <sanjeev.mudholkar@gmail.com><br>To: freeswitch-dev@lists.freeswitch.org<br>Sent: Friday, January 25, 2008
1:18:29 AM<br>Subject: [Freeswitch-dev] Call Hold ans swapping on FXS<br><br>
Hi,<br> I'am trying to use Freeswitch(with FXS) as a UA. Everything is<br>fine for a outgoing and incoming call. <br><br>1)On doing a hold (flash) on the FXS the call is put on hold <br> but there no SIP signaling (i.e re-invite is not sent to the peer).<br>
How can I force freeswitch to send a re-invite?<br><br>2) I'am not able to establish a consultation call as it is conferenced with the first<br> one as soon as the second call is connected. This also prevents<br>
from swaping between calls. The call swapping feature seems to have<br> been removed by introducing the transfer/conference feature.<br> Is there away to achieve call swapping ?<br><br>Appreciate your help.<br><br>
Thanks in advance<br>-Sanjeev<br><br>
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