Tag : codec
Before I got an log a ticket, can I just check Im not doing anything wrong?In 15.2, to install Opus:1) run `make menuselect`2) Highlight Codec Translators and press enter.3) Scroll down to codec_opus in the section labeled External4) Press enter to sel..
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to yes the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile.But in version 13.1 (I think version 11.2 also) is not working l..
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..
Use sox to downsample to 8khz (and 1 chan), and the problems should go away. While you are at it, you could use sox to convert to the target format in a single operation.The scripts that Digium uses to take Allison’s voice prompts (at 48khz) to ..