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<p class="MsoNormal">Hi,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">We have a Microsoft Lync environment for our internal phone system and are using FreeSwitch to let Lync talk with our VOIP provider. Whilst the VOIP provider properly support Lync (one of the few to do so) our NAT firewall ruins the TCP-based
SIP packets as it doesn’t forward through the SIP OPTIONS packet and Lync eventually thinks the VOIP provider isn’t there. Thus we’re using FreeSwitch to support out NAT firewall (I’ve give it our static public IP address) and have disabled the firewall device’s
SIP ALG so that it doesn’t ruin the SIP packets.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Anyway, I have this working very well except that most of the time (not all the time – maybe 90%) we just have silence for the first 3-6 seconds after each call connects. This usually means that if we call someone we miss the “Hi, thanks
for calling company XYZ. You’re speaking with ABC” and confusion results. The call counter in Lync starts counting as soon as the call connects (we’ve tested with fixed landlines in one hand and our Lync-based phone in the other) but there’s just silence.
Lync to Lync calls don’t have this problem.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I’ve tried various options such as forcing codecs (as best as I could make it anyway) to G711a since that’s all Lync supports anyway (our at least it’s happy to use this). I’ve also tried
<o:p></o:p></p>
<p class="MsoNormal"><param name="rtp-autoflush-during-bridge" value="false"/> in both legs of the FreeSwitch connections (ie to our SIP trunk provider and to Lync) without any success. I was hoping that if it was some sort of lag then at least I would see
it lag a lot in the call (and then have a definite problem to solve) but that wasn’t the case as nothing changed.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Should I do something with the proxy media settings? Reading the Wiki page at
<a href="http://wiki.freeswitch.org/wiki/Proxy_Media">http://wiki.freeswitch.org/wiki/Proxy_Media</a> seems to indicate that it really won’t make much of a difference (our CPU isn’t all that busy in the virtual machine running FreeSwitch and we only have a
handful of phones anyway). Also, I think I’d like to take advantage of some of FreeSwitch’s very powerful features in the future to take care of some of what seems trickier to do in Lync such as having our support hunt group fall back to mobile phones when
we’re out of the office (Lync does simultaneous ringing of a user’s mobile phone for direct calls but not, as far as I can tell, for hunt groups)<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I previously had the 3CX software SIP phone talking to FreeSwitch when I was first testing out the system. I’ve since gutted the FreeSwitch XML config (the “internal” / “external” SIP profiles are gone) so I can’t try the software phone
at the moment but hope to get it going again soon. This way I can better see if it’s Lync or FreeSwitch causing the problem. Before I fumble around getting that going I was hoping someone might just know the obvious thing I’m overlooking! <span style="font-family:Wingdings">J</span><o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks very much for your help.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Ian<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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