Hello all, I just set up an asterisk box (1.0.9). Its running on slackware linux 10.2 on a dual xeon 2.8 with 2 gb of ram and scsi disks.
I have 2 tdm 04b (8 fxo modules) on this box and attached there are 40 SIP peers (12 GXP 2000 and 28 budgetone 486) The problem is that when people call from the PSTN net they sometimes hear a choppy intro sound and sometimes they hear it with a low volume. Another problem is that when people dial an extension (SIP) when they call the asterisk box, sometimes the SIP peer and the caller cannot hear each other or they have a flaky audio. This basically never happens when i dial internally from SIP to SIP. The load on the machine never exceeds 0.6 The cables that connect the PSTN to the TDM cards are about 26-27 mt long but they are brand new. Can anyone help me solve my problem? Could it be also an internal QOS problem? Because when there are less peers connected everything seems better (but noe good enough). Best Regards, Fabrizio Mazzoni Macron SPA -- Fabrizio Mazzoni http://macron.com _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
