<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div>Forgot to include the list before....</div><div><br></div><div><br></div><div><br></div><div><br></div><div><div><div>On Feb 26, 2009, at 2:42 PM, Yehavi Bourvine wrote:</div><blockquote type="cite"><span class="Apple-style-span" style="color: rgb(0, 0, 0); ">2009/2/26 Zahid Mehmood <span dir="ltr"><<a href="mailto:zm23@columbia.edu">zm23@columbia.edu</a>></span><br><blockquote class="gmail_quote" style="padding-left: 1ex; margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0.8ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid; "><div style="word-wrap: break-word; ">i tested directed pickup and it worked fine in pure sip environment.. the only issues i had were with the cisco media gateway not working properly with REFER etc. Basically, instead of using openser module to keep track of the dialog, the phone A (used to pick) subscribes directly to the phoneB (ringing phone) for is dialog state, receives the notify and then acts on it. Trust me.. it would take long to get the basic working.. it just works. There will be more work if you want to implement authorization and other security measures.<div></div></div></blockquote><div> </div><blockquote class="gmail_quote" style="padding-left: 1ex; margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0.8ex; border-left-color: rgb(204, 204, 204); border-left-width: 1px; border-left-style: solid; "><div style="word-wrap: break-word; "><div><span id=""></span><font color="#ff0000">How do you do that?</font></div></div></blockquote></span></blockquote></div><br><div><br></div><div><br></div><div>I had meant to say that it "won't take long to get the basic working". </div><div><br></div><div>For testing, start with the basic opensips configuration that does not use a presence server.</div><div><br></div><div>Enable feature.11.name="group-call-pickup" and feature.12.name="directed-call-pickup" in Polycom phone configuration. After rebooting the phones, now if the phone goes off-hook, you will see a new soft key option "pickup". Pressing pickup will bring you to another screen where you enter the ringing extension and press the "Directd" soft key.</div><div><br></div><div>Pressing that would generate a subscribe to that number. Sample packet:</div><div><br></div><div><div>Directed call pickup:</div><div><br></div><div>U 2009/02/27 10:24:58.028551 192.168.12.147:5060 -> 192.168.1.50:5060</div><div>SUBSCRIBE <a href="sip:12345@">sip:12345@</a>mysiphost.com SIP/2.0.</div><div>Via: SIP/2.0/UDP 192.168.12.147;branch=z9hG4bKfdaa927d9F490C1E.</div><div>From: "Zahid" <<a href="sip:10512@">sip:10512@</a>mysiphost.com>;tag=32392D2-480E0ECB.</div><div>To: <<a href="sip:12345@">sip:12345@</a>mysiphost.com>.</div><div>CSeq: 1 SUBSCRIBE.</div><div>Call-ID: <a href="mailto:2ba849b7-7f9ea388-e763efd1@192.168.12.147">2ba849b7-7f9ea388-e763efd1@192.168.12.147</a>.</div><div>Contact: <<a href="sip:10512@">sip:10512@</a>192.168.12.147>.</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.</div><div>Event: dialog.</div><div>User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.3.0401.</div><div>Accept: application/dialog-info+xml.</div><div>Max-Forwards: 70.</div><div>Expires: 0.</div><div>Content-Length: 0.</div><div>.</div><div><br></div><div>Now, if you opensips config simply relays any "subscribe" and "notify" (or only with event: dialog) you will be able to retrieve the call ringing on "12345" from the phone "10512".</div><div><br></div><div><br></div><div>Here is a sample subscribe for "group call pickup". In this case your opensips config should detect the special username (groupcallpickuy) in the ruri and take action to build and append branches that make that group. Parallel fork to all those branches and the ringing call will be answered from your phone. If more than one phone is ringing in the group, then the first to reply will be answered.</div><div><br></div><div><br></div><div>Group call pickup:</div><div><br></div><div><div>U 2009/02/27 10:25:11.524118 192.168.1.50:5060 -> 128.59.62.26:5060</div><div>SUBSCRIBE <a href="sip:groupcallpickup@">sip:groupcallpickup@</a>mysiphost.com SIP/2.0.</div><div>Record-Route: <<a href="sip:192.168.1.50;lr=on;ftag=D0E153EE-4A592307">sip:192.168.1.50;lr=on;ftag=D0E153EE-4A592307</a>>.</div><div>Via: SIP/2.0/UDP 192.168.1.50;branch=z9hG4bK19e.5fcfad05.0.</div><div>Via: SIP/2.0/UDP 192.168.12.147;branch=z9hG4bK8314c4f96A1866BA.</div><div>From: "Zahid" <<a href="sip:10512@">sip:10512@</a>mysiphost.com>;tag=D0E153EE-4A592307.</div><div>To: <<a href="sip:groupcallpickup@">sip:groupcallpickup@</a>mysiphost.com>.</div><div>CSeq: 1 SUBSCRIBE.</div><div>Call-ID: <a href="mailto:c1f9ab73-3096f364-b277dccd@192.168.12.147">c1f9ab73-3096f364-b277dccd@192.168.12.147</a>.</div><div>Contact: <<a href="sip:10512@">sip:10512@</a>192.168.12.147>.</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.</div><div>Event: dialog.</div><div>User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.0.3.0401.</div><div>Accept: application/dialog-info+xml.</div><div>Max-Forwards: 69.</div><div>Expires: 0.</div><div>Content-Length: 0.</div><div>.</div><div><br></div><div>Once presence_dialoginfo and pua_dialoginfo become stable, you can start using that to process all the subscribe/notify messages instead of relying on the endpoints.</div><div><br></div><div>Hope this helps.</div><div><br></div><div>-- </div><div>Zahid</div></div></div></div><div><br></div><div>On Feb 24, 2009, at 7:48 AM, Iņaki Baz Castillo wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div>2009/2/24 Yehavi Bourvine <<a href="mailto:yehavi.bourvine@gmail.com">yehavi.bourvine@gmail.com</a>>:<br><blockquote type="cite">Hello,<br></blockquote><blockquote type="cite"><br></blockquote><blockquote type="cite"> I am in the process of duplicating my Asterisk system into OpenSIPS in<br></blockquote><blockquote type="cite">order to allow for a future growth. I need to do directed pickup when<br></blockquote><blockquote type="cite">another extension rings. How do I do that? (assuming I know who wants to<br></blockquote><blockquote type="cite">pickup what).<br></blockquote><br>OpenSIPS has a new module presence_dialoginfo and pua_dialoginfo,<br>implementing RFC 4235 which allows call pick-up and so, but it doesn't<br>work very well for now.<br><br>If Asterisk is in the middle of the calls then you can remain using<br>Asterisk PickUp (very poor anyway, but it "works").<br><br>-- <br>Iņaki Baz Castillo<br><<a href="mailto:ibc@aliax.net">ibc@aliax.net</a>><br><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>http://lists.opensips.org/cgi-bin/mailman/listinfo/users<br></div></blockquote></div><br></body></html>