[Asterisk-Users] One sided call

[Adrian Carter]

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).



Bruce Reeves wrote:
> 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.
>
> On 5/11/06, *Adrian Carter* <adrian at lei.net.au 
> <mailto:adrian at lei.net.au>> wrote:
>
>     I had a very similar issue just today with some Linksys SPA-941's...
>     Of a collection  15, 5 of them had consistent 'one-sided-audio' on
>     INBOUND calls, but worked fine on OUTBOUND calls.
>
>     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.
>
>     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..
>
>     Regardless, after performing these steps everything works 100% now.
>
>
>
>     Woodoo People .pGa! wrote:
>>     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
>>      
>>       
>>>     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:
>>>         
>>       
>
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>
> -- 
> Bruce
> Nortex Networks
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