Hi! From my perspective I know that maintaining a SIP stack requires *A LOT* of effort, so I understand that a project can’t maintain two of them.
I suggest that a working group is created for the transition and that the first task is to compare the functionality. Last time I checked the functionality *I need* (but maybe not everyone else) was non-existing in PJSIP so I could not use it. It may have changed since then. I think the goal has to be to gradually phase out the ugly code in chan_sip and celebrate the day it’s gone, but make sure we don’t leave functionality (and users) behind and have good guidelines for the transition. I still think we should totally rewrite how chan_pjsip is configured. That concept is very far away from other SIP implementations. But that’s my personal opinion from a small cold corner of the world, using Asterisk in non-PBX ways as large scale media and feature servers. Executive summary: Create a working group that maintains the feature gap, makes sure it’s going away and also makes sure that we have enough material that explains the gold that hides in chan_pjsip! /O -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
