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<DIV>Hi Rupa</DIV>
<DIV> </DIV>
<DIV>I've got similar settings here, but I can't make them work, nothing happens when I press R:</DIV>
<DIV> </DIV>
<DIV>The settings are</DIV>
<DIV>DTMP over VoIP connections: Send settings [ ]Auto [X] Audio [ ] RFC 2833 [X] SIP Info</DIV>
<DIV> </DIV>
<DIV>Call Transfer</DIV>
<DIV>Use the R key to initiate call ( ) Yes (X) No</DIV>
<DIV>transfer with the SIP Refer Method</DIV>
<DIV> </DIV>
<DIV>Transfer Call by On-Hook (X) Yes ( ) No</DIV>
<DIV> </DIV>
<DIV>Derive Target address ( ) from SIP URL (X) from SIP Contact Header</DIV>
<DIV> </DIV>
<DIV>Find Target address automatically ( ) Yes (X) No</DIV>
<DIV> </DIV>
<DIV>Hook Flash (R-key)</DIV>
<DIV> </DIV>
<DIV>Application Type: dtmf-relay</DIV>
<DIV>Application Signal: 16</DIV>
<DIV> </DIV>
<DIV>That's all the settings, I had to set 'send settings' from auto to audio+SIP Info otherwise the R-Flash settings were disabled.</DIV>
<DIV><BR> </DIV>
<DIV>But all in all, I think a *1 or something during the call would be a better method, because I should be able to make that work on al types of phones, right?</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>
<DIV>Best regards</DIV>
<DIV>Fribse</DIV>
<DIV> </DIV>
<DIV>/Running Freeswitch on pfSense running on a VMWare ESXi 4.0</DIV></DIV>
<DIV><BR>>>> 11-02-2010 kl. 14:28 skrev Rupa Schomaker <rupa@rupa.com> i meddelelsen <b39273531002110528p42218d07q95e97485139315f1@mail.gmail.com>:<BR></DIV>
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<DIV style="BORDER-LEFT: #050505 1px solid; PADDING-LEFT: 7px">My Siemens A580 has options for controlling the R key. It seems that you can either have it setup for transfer or as a hook flash. Default is as a transfer key.
<DIV><BR></DIV>
<DIV>I haven't succeeded in getting it to work for transfer and it is wayyyyy down low on my list of things to do with the phone.</DIV>
<DIV><BR></DIV>
<DIV>From the web UI:</DIV>
<DIV><BR></DIV>
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<H2 class=first>Call Transfer</H2></TD>
<TD></TD></TR>
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<TD class=label>Use the R key to initiate call transfer with the SIP Refer method.:</TD>
<TD class=rightSide><INPUT class=radioButton value=1 CHECKED type=radio name=ct_use_rkey>Yes <INPUT class=radioButton value=0 type=radio name=ct_use_rkey>No </TD></TR>
<TR>
<TD class=label>Transfer Call by On-Hook:</TD>
<TD class=rightSide><INPUT class=radioButton value=1 type=radio name=ct_after_on_hook>Yes <INPUT class=radioButton value=0 CHECKED type=radio name=ct_after_on_hook>No </TD></TR>
<TR>
<TD class=label>Derive target address:</TD>
<TD class=rightSide><INPUT class=radioButton value=0 type=radio name=ct_refer_to>from SIP URL <INPUT class=radioButton value=1 CHECKED type=radio name=ct_refer_to>from SIP contact header </TD></TR>
<TR>
<TD class=label>Find target addr. automatically:</TD>
<TD class=rightSide><INPUT class=radioButton value=1 type=radio name=ct_refer_auto>Yes <INPUT class=radioButton value=0 CHECKED type=radio name=ct_refer_auto>No </TD></TR>
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<TD style="PADDING-BOTTOM: 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; PADDING-TOP: 0px"><IMG border=0 alt="" src="http://a580-ip.rupa.com/images/spacer.gif" width=200 height=1></TD>
<TD style="PADDING-BOTTOM: 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; PADDING-TOP: 0px"><IMG border=0 alt="" src="http://a580-ip.rupa.com/images/spacer.gif" width=360 height=1></TD></TR></TBODY></TABLE>
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<H2 class=first>Hook Flash (R-key)</H2></TD>
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<TD class=rightSide>R key settings are disabled because the R key is being used for call transfer. </TD></TR></TBODY></TABLE></DIV><BR>
<DIV class=gmail_quote>On Wed, Feb 10, 2010 at 11:45 PM, mailinglist <SPAN dir=ltr><<A href="mailto:mailinglist@fribert.dk">mailinglist@fribert.dk</A>></SPAN> wrote:<BR>
<BLOCKQUOTE style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class=gmail_quote>
<DIV style="MARGIN: 4px 4px 1px; FONT: 10pt Segoe UI">
<DIV>Sorry for the repost, but the previous thread just died :-)</DIV>
<DIV></DIV>
<DIV>I'm trying to get the possibility of transfering an incoming call from one extension to another, and give the possibility of turning it into a conference.</DIV>
<DIV>I don't have a 'transfer' button.</DIV>
<DIV>I do have an 'R' button on the Siemens handsets, and a 'Flash' button on the Sipura. The 'Flash' button gives me a new dialtone, gives the caller MOH, and then I can dial the new extension, and transfer the call, but not create a conference.</DIV>
<DIV>But the Siemens handset does not have a 'flash', and pressing the R doesn't do anything.</DIV>
<DIV></DIV>
<DIV>It might be two different features 'transfer' and 'conference'...</DIV>
<DIV><BR>But I thought that using the bind_meta_app would accomplish both.</DIV>
<DIV></DIV>
<DIV>It's on an incoming call from the outside.</DIV>
<DIV>So the situation:</DIV>
<DIV>The Public folder has an entry that matches the dialed number, and does a transfer to 8202.</DIV>
<DIV>Then the dialplan matches the 8202 with a group, and the phone rings.</DIV>
<DIV>Somebody picks it up, finds out that it needs to be transferred to another extension, or transferred to a conference with a second extension.</DIV>
<DIV>How do I construct that?</DIV>
<DIV></DIV>
<DIV></DIV>
<DIV></DIV>
<DIV>Best regards</DIV>
<DIV>Fribse</DIV>
<DIV></DIV>
<DIV>/Running Freeswitch on pfSense running on a VMWare ESXi 4.0</DIV></DIV><BR>_______________________________________________<BR>FreeSWITCH-users mailing list<BR><A href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</A><BR><A href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target=_blank>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><BR>UNSUBSCRIBE:<A href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target=_blank>http://lists.freeswitch.org/mailman/options/freeswitch-users</A><BR><A href="http://www.freeswitch.org" target=_blank>http://www.freeswitch.org</A><BR><BR></BLOCKQUOTE></DIV><BR><BR clear=all><BR>-- <BR>-Rupa<BR></DIV></DIV></TD></TR></TBODY></TABLE></BODY></HTML>