<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:'times new roman', 'new york', times, serif;font-size:10pt"><div>Hi Mike</div><div><br></div><div>let me make it clear :)</div><div><br></div><div>ip phone1 (ext 1000) registers to SIP server</div><div>ip phone2 (ext 1002) registers to SIP server</div><div>FS (ext 1003) register to SIP server</div><div><br></div><div>the media proxy and sip server are in the same server</div><div><br></div><div>the signaling flow :</div><div> ext1000 ---> INVITE ---> SIP server ---> INVITE---> ext 1002</div><div><br></div><div>the media flow:</div><div>ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002</div><div><br></div><div>so the signaling for FS when i call 1003</div><div>step1: <span class="Apple-tab-span" style="white-space:pre"> </span>ext1000--->INVITE--->Sip
server---->INVITE---->FS</div><div>step2: <span class="Apple-tab-span" style="white-space:pre"> </span>ext1000 press 1002, FS will transfer call to SIP server</div><div>signaling flow for step 2:</div><div>1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold</div><div>2. FS --->INVITE---> SIP server----> INVITE--->ext1002</div><div>when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer </div><div>3. ext1000 connected to ext1002</div><div><br></div><div>media flow:</div><div>step1:<span class="Apple-tab-span" style="white-space:pre"> </span>ext1000----RTP---media proxy----RTP---FS</div><div>step2:</div><div>1. FS -----no RTP---media proxy----no RTp----ext1000</div><div>2. FS----RTP----media proxy---ext1002</div><div>3 ext1000----media proxy----ext1002</div><div><br></div><div><br></div><div>Thank
you</div><div><br></div><div><br></div><div style="font-family:times new roman, new york, times, serif;font-size:10pt"><br><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">Từ:</span></b> Michael Jerris <mike@jerris.com><br><b><span style="font-weight: bold;">Đến:</span></b> freeswitch-users@lists.freeswitch.org<br><b><span style="font-weight: bold;">Gửi ngày:</span></b> 11:23:25, Thứ Hai, 26 tháng 4 2010 <br><b><span style="font-weight: bold;">Chủ đề:</span></b> Re: [Freeswitch-users] need help on IVR<br></font><br>I meant this literally. People always use the term "sip server" and I never understand what they are trying to say. What exactly is a sip server? I think you have your terms all completely confused in your explanation below or I am just being really dense. This cisco server is a sip server
and a media proxy? Can you try to be very clear on what exactly these things are and what you are trying to do?<div><br></div><div>Mike</div><div><br><div><div>On Apr 25, 2010, at 10:44 PM, false wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse:separate;font-family:Helvetica;font-size:medium;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;orphans:2;text-indent:0px;text-transform:none;white-space:normal;widows:2;word-spacing:0px;"><div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;font-family:times, serif;font-size:10pt;"><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">Hi Michael Jerris</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div
style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">my sip server is cisco sip server</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">could you guide me some clue or show the same config for the case</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">Thank you</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;font-family:times, serif;font-size:10pt;">Ha`<br><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;font-family:times, serif;font-size:12pt;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight:bold;">Từ:</span></b><span class="Apple-converted-space"> </span>Michael Jerris <<a rel="nofollow" ymailto="mailto:mike@jerris.com" target="_blank"
href="mailto:mike@jerris.com">mike@jerris.com</a>><br><b><span style="font-weight:bold;">Đến:</span></b><span class="Apple-converted-space"> </span><a rel="nofollow" ymailto="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br><b><span style="font-weight:bold;">Gửi ngày:</span></b><span class="Apple-converted-space"> </span>1:08:26, Chủ nhật, 25 tháng 4 2010<span class="Apple-converted-space"> </span><br><b><span style="font-weight:bold;">Chủ đề:</span></b><span class="Apple-converted-space"> </span>Re: [Freeswitch-users] need help on IVR<br></font><br>what is a sip server? Yes, you can do pretty much anything like this.<div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><div
style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">On Apr 19, 2010, at 12:08 AM, false wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse:separate;font-family:Helvetica;font-size:medium;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;orphans:2;text-indent:0px;text-transform:none;white-space:normal;widows:2;word-spacing:0px;"><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;font-family:times, serif;font-size:10pt;"><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">Hi all</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">my network
topology:</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">endpoint 1(100)-----sip server ---IVR(Freeswitch)</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><span class="Apple-tab-span" style="white-space:pre;"> </span>|</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><span class="Apple-tab-span" style="white-space:pre;"> </span>|<span class="Apple-tab-span" style="white-space:pre;"> </span></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><span class="Apple-tab-span" style="white-space:pre;"><span class="Apple-tab-span" style="white-space:pre;"> </span>e</span>ndpoint2(101)</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div
style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">endpoint1 + endpoint2 are registered to sip server</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">Freeswitch is regsitered to sip server with 103</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR </div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">and RTP from endpoint 1 --> media proxy---> FS</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">and RTP from endpoint1--> media proxy
-->endpoint1, RTP will not go through the FS after FS transfer call to 101</div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;"><br></div><div style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0px;">can FS do it</div></div></div></span></blockquote></div></div></div></div></div></div></span></blockquote></div><br></div></div></div><div style="position:fixed"></div>
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