Summary:	ASTERISK-27834: PJSIP - Implement SIP_CODEC_INBOUND
Reporter:	Niklas Larsson (pnlarsson)	Labels:	pjsip
Date Opened:	2018-05-03 02:14:07	Date Closed:
Priority:	Minor	Regression?	No
Status:	Open/New	Components:	Channels/chan_pjsip
Versions:	GIT	Frequency of Occurrence
Related Issues:	must be completed before resolving ASTERISK-27309 Feature Parity with chan_sip
Environment:		Attachments:
Description:	Missing feature to chan_sip - ability to set codec on inbound call. PJSIP_MEDIA_OFFER sets the codecs on the outbound call.
Comments:	By: Asterisk Team (asteriskteam) 2018-05-03 02:14:08.531-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].