i use predective with ratio 2.0 or same time i use same time the adapt hard limit with 2.0 dial level too
and then i will explain same thing about my provider
my provider use a sip account with g729 as codec
and i my server i' don't have any g729 actived or buyed from digium and also i have see that asterisk 1.2 don't allow the g729 since they are asterisk 1.4 so here i have try this dialplan in my trunk add with dialplan:
[trunk_1]
type=peer
disallow=all
allow=alaw
allow=ulaw
username=xxxxxxx
fromuser=xxxxxxxx
secret=xxxxxx
host=xx.xx.xx.xx
TRUNKX=SIP/trunk_1
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Hangup
finally i have any probleme with predective or with manual dial just
when i have try vicibox with 4 go of ram and with dual core2duo as config to run 8 seats or more .
so here i receive a bad sound quality from the other side and same time i heard them and some time me too i get a bad sound .
so just i seach a answer for my probleme it is a codec probleme coming from this g729 or it is a
bandwidth probleme "for information i have a LS internet conection with 512 kb/s as speed .
so here i don't know what i need to do if i buy a licence from duguim can i resolve this probleme ;or it is other probleme .
maybe bcause i use also the ulaw and alaw and g711 and gsm as default codec found in asterisk 1.2 and this g711 it use 64kb/s as
bandwidth.
thanks for all answer