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Hello, I switched from my PSTN provider to a voip provider. (Voicedata in the Netherlands) >From the moment i switched all inbound calls are terminated after aproximatly 1 minute. The provider tells me it's not their issue since I have no other configuration than all their other users. What can I do. I removed all asterisk functionality by forwarding the inboud call directly to a local phone ; Inbound voicedata context ; [from-voicedata] exten => ${VOICE...

...) when i try to make call form one pstn phone to other trough asterisk or when i make call from software h.323 client trough asterisk and this gateway to pstn i have the problem with voice quality. The side that initiated call can be heared clearly, but the side that answered the call can be heared aproximatly at half. E.g. on this side everything is heared with some cuts. I runed asterisk -vvvc and during the call it constantly printning: WARNIGN .. File chan_oh323.c, Line 761 (oh323_read) H323:20016 Requested read buffer size is too long (2560) ! and next line message about truncating buffer. I'...

...bscribe: <http://lists.samba.org/mailman/listinfo/samba>, <mailto:samba-request@lists.samba.org?subject=unsubscribe> List-Archive: http://lists.samba.org/pipermail/samba/ First off a little bit of background... we killed our last operational NT server back 4 months ago. We run a WAN of aproximatly 30 offices each containing a samba 2.0.x server... Each remote server operates under it's own workgroup (specific to that office and provides domain logon services to win9x clients. These remote sambas then forward the password request via Security = server Password Server = 192.16...