Well, i think i've solved the problem by myself :-) I had to change a line in chan_capi_pvt.h:
/* was : 130 bytes Alaw = 16.25 ms audio not suitable for VoIP */ /* now : 160 bytes Alaw = 20 ms audio */ /* you can tune this to your need. higher value == more latency */ #define AST_CAPI_MAX_B3_BLOCK_SIZE 160 Putting the AST_CAPI_MAX_B3_BLOCK_SIZE to 130 (16.25ms audio) solved the problem. I forgot to mention that i'm using Snom105 phones. It seems that with GS BT101 with Ilbc firmware the value 160 works fine, but with snom it introduces an ugly distortion and choppy audio. Recompiled using 130 as value, and the sound now is just really fine. A little question to Kapejod if he reads this... Is it possibile to put an even littler value here? I've tried to use 80 (= 10ms audio) but it makes impossible to start * cause it can't load chan_capi.so module. Regards, -- Stefano _______________________________________________ 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
