<div dir="ltr">Dear Bogdan,<br><br>Do you mean doing something like that?<br><br>if(!cr_route("default", "0", "$rU", "$rU", "call_id")){<br> sl_send_reply("403", "Not allowed");<br>
} else {<br> # In cas of failure, re-route the request<br> t_on_failure("1");<br><br>force_rtp_proxy();<br> t_relay();<br>route(2) ;<br>}<br><br><br>Please let me know how can I force it on route2 for the 200OK reply<br>
<br>Thanks Bogdan<br><br><br><div class="gmail_quote">On Fri, Feb 27, 2009 at 11:54 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi Michel,<br>
<br>
You have to call twice force_rtp_proxy() in order to complete the RTP session (and have RTP flowing). First for the INVITE (as you already do) and second for the 200 OK reply.<br>
<br>
So install a onreply_route and if the reply is 200 OK, call again force_rtp_proxy.<br>
<br>
Regards,<br>
Bogdan<br>
<br>
michel freiha wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">
Dear All,<br>
<br>
I need to make all my rtp traffic through OpenSips to pass through rtp proxy...I have the following route:<br>
<br>
<br>
if(!cr_route("default", "0", "$rU", "$rU", "call_id")){<br>
sl_send_reply("403", "Not allowed");<br>
} else {<br>
# In cas of failure, re-route the request<br>
t_on_failure("1");<br>
force_rtp_proxy();<br>
t_relay();<br>
}<br>
The call is working fine but with no audio...How can i fix this issue in order to have 2 way audio through rtpproxy?<br>
<br>
Regards<br></div></div>
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</blockquote>
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</blockquote></div><br></div>