[asterisk-users] Why doesn't Asterisk try to prevent transcoding

CDR venefax at gmail.com
Sun Dec 15 14:55:48 CST 2013

I have had the issue for years. The problem is that Asterisk
developers are removed from the business. We desperately need simple
way to eliminate transcoding when unnecessary. Transcoding brings a
server to its knees. It is a very simple new setting in sip.conf
I vote for this new feature. However, I don't have the expertise to
write  a patch. I would say that only Digium developers could attempt
to do this without disrupting the code too much. I also tried to
migrate to PJSIP, but had to go back when I realized there was no
channel variable contaning the inbound IP address. In general, any
channel hast to provide the information to the dialplan, somehow,
otherwise we cannot do business. I hope the PJSIP integration matures

More information about the asterisk-users mailing list