[Asterisk-Users] Re: One sided call

[Bruce Reeves]

I have contiuned to fight this problem all day, and still have not found a
solution. I did get DTMF tones to the other end, but no voice. Any tips on
where to look?

On 5/10/06, Bruce Reeves <asterisk at nortex-networks.com> wrote:
>
> 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.confand 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:
>
> NOTICE[24975]: rtp.c:330 process_rfc3389: Comfort noise support incomplete
> in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:
> 192.168.10.56
>
> But I have silence suppresion off on the linksys phone.
>
> --
> Bruce
> Nortex Networks
>



-- 
Bruce
Nortex Networks
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