Is there a public download site for Linksys/Sipura firmwares? I found
nothing on Linksys site. I'm currently running 4.1.10(e) on my SPA-942.<br><br><div><span class="gmail_quote">On 5/11/06, <b class="gmail_sendername">Adrian Carter</b> <<a href="mailto:adrian@lei.net.au">adrian@lei.net.au
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div>
<div bgcolor="#ffffff" text="#000000">
I had a very similar issue just today with some Linksys SPA-941's...<br>
Of a collection 15, 5 of them had consistent 'one-sided-audio' on
INBOUND calls, but worked fine on OUTBOUND calls.<br>
<br>
In the end, a flash upgrade to 4.1.12(a), a factory reset, and a
reconfig fixed the problem... Same settings went back into the phone
again so I have no idea what this fixes as the phones where already on
4.1.12(a) without the factory reset and it still didn't clear it up. It
wasn't untill the factory reset and reconfig that they finally worked.<br>
<br>
It should be noted I broke the seal on all these phones from the box,
so really, one would have expected consistent behaviour across all the
phones.. <br>
<br>
Regardless, after performing these steps everything works 100% now. </div><div><span class="q"><br>
<br>
<br>
Woodoo People .pGa! wrote:
<blockquote cite="http://mid20060511073030.GA24689@Shadow.microsystem.hu" type="cite">
<pre>Hi! I found, that there is 4 options for nat:<br>-no<br>-never<br>-yes<br>-always<br><br>no and never is ok<br>but sometimes yes, and sometimes always worked for me :-o<br> <br> </pre>
<blockquote type="cite">
<pre>I am having problem diagnosing a call problem. On both a Cisco phone and a<br>Linksys 942 I am only getting one side of the call when connected over a WAN<br>link or internet connection. I have set nat=yes and qualify in
sip.conf and<br>the phone registers fine. I can hear the other end, but they do not hear<br>anything, no voice or dtmf. I found a tip about changing the RTP rate from<br>.03 to .02 on Sipura phones to match Asterisk rate and did that. I also made
<br>sure the RTP range for the phone and the server was set to 10000 thru 20000.<br>These phones work fine when on the same subnet as the server. The server<br>shows the following message:<br> </pre>
</blockquote>
<pre> </pre>
</blockquote>
</span></div><div></div>
</div><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://Easynews.com" target="_blank">Easynews.com</a> --
<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">
http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br></blockquote></div><br><br clear="all"><br>-- <br>Bruce<br>Nortex Networks