Hi: my ITSP use G.711 with VAD and it can not change the settings. I was using Asterisk 13 but the voice quality is not very good. I dont know if asterisk 16 is good enough to handle this kind of situation? or I still need FreeSWITCH in front of Aster..
Author : d tbsky
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hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didnt work with pjsip in asterisk12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems ..
Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDIuse alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to al..