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I actually got it off the voip-info.org's Wiki search for SPA-941 .. I
found that at the believe it or not sipura.com website - 4.1.12(a).<br>
<br>
<br>
<br>
Bruce Reeves wrote:
<blockquote
cite="mid1941adec0605110832n3a1e12c6i68b74275c8316696@mail.gmail.com"
type="cite">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:
-no
-never
-yes
-always
no and never is ok
but sometimes yes, and sometimes always worked for me :-o
</pre>
<blockquote type="cite">
<pre>I am having problem diagnosing a call problem. On both a Cisco phone and a
Linksys 942 I am only getting one side of the call when connected over a WAN
link or internet connection. I have set nat=yes and qualify in
sip.conf and
the phone registers fine. I can hear the other end, but they do not hear
anything, no voice or dtmf. I found a tip about changing the RTP rate from
.03 to .02 on Sipura phones to match Asterisk rate and did that. I also made
sure the RTP range for the phone and the server was set to 10000 thru 20000.
These phones work fine when on the same subnet as the server. The server
shows the following message:
</pre>
</blockquote>
<pre> </pre>
</blockquote>
</span></div>
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<br>
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</blockquote>
</div>
<br>
<br clear="all">
<br>
-- <br>
Bruce<br>
Nortex Networks
<pre wrap=""><hr size="4" width="90%">
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</pre>
</blockquote>
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