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I have a server located on the internet running opensips and asterisk. When registering directly to asterisk I can perform echo tests and make calls.<BR>
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If I register to Opensips and use the load_balance there is one way audio. I can hear sounds coming from the asterisk server but sound from the soft phone does not reach asterisk. I can confirm this when looking at a rtp debug on asterisk.<BR>
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I can see that traffic is passing from the soft phone when performing a wire shark trace to the server and it also shows that some RTP packet are being passed out and back into my local address. This does not happen if I register directly to asterisk.<BR>
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Any advice you can offer would be appreciated.<BR>
<BR>
Opensips shouldn't effect the RTP if it only load balances?<BR>
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Thanks,<BR>
<BR>Ross<BR> <br /><hr />Did you know you can get Messenger on your mobile? <a href='http://clk.atdmt.com/UKM/go/174426567/direct/01/' target='_new'>Learn more.</a></body>
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