asterisk users - Jan 2006 - Re: Asterisk-Users Digest, Vol 18, Issue 158

Hi,

I have already set canreinvite=no in the sip.conf and also used the
NAT=yes.  But the funny thing that was in one case the user call and it
wasn't working (one way audio as described) using an online dialer and then
tried again using X-lite it was working.  Then hanged up and tried X-lite
again, it was not working.  The second call was only a few seconds apart.
Moving back to the online dialer, it wasn't working either.  So it is just
very strange to me how this happened and i was think maybe it was the RTP
negiotation.  Do you have any ideas?

Regards,
Kengie

>Few people, or no one, will take the time to see all the debug.
>The key here is that the RTP port and IP negotiated in the SDP message
>sent by asterisk to each party, should be "visible" for the party.
A
>common error is Asterisk sending in SDP a private IP address to a
>public UA, so the public UA will attempt to send RTP audio to a
>private IP, never reaching the Asterisk Server. Check voip-info.org
>about RTP issues with NAT, check the option canreinvite in sip.conf,
>put canreinvite=no , may be that will help. If you have one of the UA
>behind a NAT, use nat=yes
>regards
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