I have a user that has been able to conference in customers, three lines on his phones, but since a reboot of the Asterisk server is no longer able to add in a third call. Any ideas? Thanks.
Is he using the the phone’s 3 party calling, or Asterisk features.conf three party calling?
What does the “sip set debug on” output show?
I figured out the issue. I just needed to add more codecs to the allow line.
g729&ulaw&alaw