diff options
| author | zuul <zuul@gerrit.asterisk.org> | 2016-09-09 13:56:16 -0500 |
|---|---|---|
| committer | Gerrit Code Review <gerrit2@gerrit.digium.api> | 2016-09-09 13:56:16 -0500 |
| commit | 9d54dd04bbdad6849aee77536ab12c5fa6620680 (patch) | |
| tree | 16042126c4d0eac8d8308a8d18b3b4f776c12f41 /CHANGES | |
| parent | 901e612739e6067c4d51656f35c49f005534f1de (diff) | |
| parent | 2a50c2910144e1b4095d171b1386fd5ebb0c5b5a (diff) | |
Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 7 |
1 files changed, 7 insertions, 0 deletions
@@ -20,6 +20,13 @@ chan_sip a dialplan that dials with it enabled initially and if it fails fall back to without. +res_pjsip +------------------ + * Added endpoint configuration parameter "preferred_codec_only". + This allow asterisk response to a SIP invite with the single most + preferred codec rather than advertising all joint codec capabilities. + This limits the other side's codec choice to exactly what we prefer. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ---------- ------------------------------------------------------------------------------ |