ok, thanks, does rtp*timeout work if i have canreinvite=yes ? since rtp traffic is not passing thought asterisk, or i have to put canreinvite=no ?
slds. > rtp*timeout for sip peers is not a fix but a > workaround. > Try to set both values and reload sip. > Then when you witness what you posted try doing a > "core show channels". You can then try to "soft > hangup" a stuck channel or wait for the rtp*timeouts. > > > > > ____________________________________________________________________________________ > Be a better friend, newshound, and > know-it-all with Yahoo! Mobile. Try it now. > http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-600-5490 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:[EMAIL PROTECTED] Email/MSN: [EMAIL PROTECTED] http://www.tiendalinux.com/ Bogota, Colombia _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users