Is the SIP phone behind NAT? That's one of the common reasons for one way audio. You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity. The SIP port ranges should be something like:
<br><br>SIP: 5060-5061<br>RTP: 10000-20000<br><br>Kyle<br><br><div><span class="gmail_quote">On 4/1/06, <b class="gmail_sendername">Il Neofita</b> <<a href="mailto:asteriskmail@gmail.com">asteriskmail@gmail.com</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 style="direction: ltr;">Hi,<br>I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing.
<br>The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP.
<br>Any thoughts?
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