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