<div dir="rtl"><div dir="ltr">I skimmed through the document and it seems that FreeSwitch can be tailored to comply with most of the requirements by the end user. There are 3 issues which I am not sure about:</div>
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<div dir="ltr">Section 12.1 states that call transfer should be done with INVITE/Re-INVITE and not by REFER. From the SIP traces I've done I see that FS uses REFER to transfer a call.<br>Section 14.1: Does FS accepts INVITE with no SDP inside?</div>
<div dir="ltr">Section 14.3: Does FS supports RFC-4733? This section allows also RFC-2833 for those who do not support 4733.</div>
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<div dir="ltr"> Thanks, __Yehavi:<br></div>
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<div>2011/3/22 Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>></span></div>
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<div class="im">On Mon, Mar 21, 2011 at 9:25 PM, Steve Underwood <span dir="ltr"><<a href="mailto:steveu@coppice.org" target="_blank">steveu@coppice.org</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">I'm not clear how you can get a PBX SIP CONNECT approved. A lot of the<br>document comes down to how you configure and use the system. Obviously,<br>
a product could present a roadblock that prevents SIP CONNECT compliance<br>in a working setup, but I doubt that many products would do that.<br><br>Characterising the SIP forum as a who's who of retarded SIP providers<br>
and PBXes is a little unfair. Practically everyone in the VoIP business<br>is in that list. Capability certainly doesn't look like a prerequisite,<br>though. :-\<br></blockquote>
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<div>Hehe, true enough. Perhaps I was a little harsh. Still, you are quite right about the compliance test being incredibly subjective and capability not being a prerequisite. Sonus and ShoreTel are not exactly known for their SIP interop features.</div>
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<div>-MC</div>
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