We've been using SIP with Asterisk for a couple of years now, and it's
generally worked fine. However we're now trying to use a more
complicated codec setup, and I've hit a problem with how codecs are
selected that I can't get around.
For a simple configuration:
XLite > GSM > Asterisk
where GS
Jamie Neil wrote:
Is this a bug, by design or is it a problem specific to Xlite?
Neither. Your analysis is correct. When the SIP INVITE comes in, and the
user has not been asked to authenticate yet, the codec settings used are
the "global" ones from the [general] section of sip.conf. This is
nec
