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<P> </P>
<P>Thanks.</P>
<P> </P>
<P>I tried your suggestion, and it make no use. </P>
<P> </P>
<P> --- </P>
<P> Best regards,</P>
<P> Qiao Yuansong</P>
<P> mailto: qys@iscas.ac.cn</P>
<P> </P>
<P> Friday, April 15, 2005, 10:21:16 AM, you wrote:</P>
<P> </P>
<P> <SPAN class=RVTS6>> I'm Andrew.</SPAN></P>
<P> </P>
<P> <SPAN class=RVTS6>> On April 14, 2005 10:01 pm, Qiao Yuansong wrote:</SPAN></P>
<P> <SPAN class=RVTS6>>> My asterisk box and sip phone are not behind a nat, the sip phone and</SPAN></P>
<P> <SPAN class=RVTS6>>> asterisk box are connected by LAN, so the delay is not caused by network</SPAN></P>
<P> <SPAN class=RVTS6>>> congestion, and furthermore, there is no delay from sip to pstn.</SPAN></P>
<P> <SPAN class=RVTS6>>></SPAN></P>
<P> <SPAN class=RVTS6>>> [sip phone]------LAN------[Asterisk with X100P]------[PSTN]</SPAN></P>
<P> <SPAN class=RVTS6>>> sip to pstn (no delay)</SPAN></P>
<P> <SPAN class=RVTS6>>> pstn to sip (half or one second delay)</SPAN></P>
<P> </P>
<P> <SPAN class=RVTS6>> This doesn't make any sense; the streams are identical. Are different codecs</SPAN></P>
<P> <SPAN class=RVTS6>> being negotiated when the call origination is one side then the other?</SPAN></P>
<P> </P>
<P> <SPAN class=RVTS6>> put</SPAN></P>
<P> </P>
<P> <SPAN class=RVTS6>> disallow=all</SPAN></P>
<P> <SPAN class=RVTS6>> allow=ulaw</SPAN></P>
<P> </P>
<P> <SPAN class=RVTS6>> in sip.conf, under [general] and comment out all other allow/disallow lines.</SPAN></P>
<P> <SPAN class=RVTS6>> Restart asterisk and try again. Something basic is not right.</SPAN></P>
<P> </P>
<P> <SPAN class=RVTS6>> -A.</SPAN></P>
<P> <SPAN class=RVTS6>> _______________________________________________</SPAN></P>
<P> <SPAN class=RVTS6>> Asterisk-Users mailing list</SPAN></P>
<P> <SPAN class=RVTS6>> Asterisk-Users@lists.digium.com</SPAN></P>
<P> <SPAN class=RVTS6>> http://lists.digium.com/mailman/listinfo/asterisk-users</SPAN></P>
<P> <SPAN class=RVTS6>> To UNSUBSCRIBE or update options visit:</SPAN></P>
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