[asterisk-users] RTP (voice) issue. STUN server

[Gopalakrishnan A.N]

Try something like this,


[general]
localnet=192.168.0.0/255.255.0.0 ; or your subnet
externip=x.x.x.x               ; use your address

[YOURREMOTEPEER]               ; your peer's name
nat=yes
qualify=yes                    ; Force keepalives



On Thu, Feb 24, 2011 at 7:12 PM, Oleg Botvinkin <olegbo at gmail.com> wrote:

> Hi,all
> I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are
> opened, externip is configured in sip.conf. I think, that all relevant
> configurations are checked. But, no voice hear between local and remote
> extension. What I need to check, configure in router and PBX for resolving
> this issue ?
> How I can to install and configure STUN server ?
> Thanks,
> Oleg
> .
>
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-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:saigop at gtalk2voip.com
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