Try to run rtpproxy on private ip not on local 127.0.0.1<br><br><div class="gmail_quote">On Tue, Jun 28, 2011 at 11:23 AM, MingHon <span dir="ltr">&lt;<a href="mailto:gminghon@gmail.com">gminghon@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi,<div><br></div><div>i fixed the audio issue for 102 to 103 vice versa.</div><div><br></div>
<div>by fixing the canreinvite in asterisk.</div><div><br></div><div>from uac the rtp packet will route to kamailio den forward to asterisk.</div>
<div><br></div><div>can we bypass the rtp packet going to asterisk?</div><div><br></div><div>and here is the update for uac 101 issue.</div><div><br></div><div>when 101 call to voicemail or 102/103 there is no audio.</div>

<div><br></div><div>in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.</div><div><br></div><div>but if 102/103 call to 101 both uac got audio.</div><div><br></div><div>i realize this is because 101 is the first uac registered before 102/103 and because it did not have the received: field in ul show.</div>

<div><br></div><div>please adv.</div><div><br></div><div>-- <br>Regards,<br><br>MingHon<br>
</div>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Regards,<br><br>Chandrakant Solanki<br>