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..