We are able to make multiple inboud/outbound calls at the same time, but when setting up an conference, it works with 1 user. When a second users dials in, it gets a busy signal.
Does the SIP trunk you use, have a DID attached to it, so callers from the PSTN can dial in? If this is the case, how many channels does your DID support?
Are you directing the incoming call to an extension and then transferring the call to a conference? If this is the case, does your extension support CallWaiting? (i.e. have you dialed *70 at least once from that extension?).
Are you directing the incoming call to an extension and then transferring the call to a conference? If this is the case, does your extension support CallWaiting? (i.e. have you dialed *70 at least once from that extension?).
We have not dialed *70 on that specific extension. Other extensions work, so that sould be the problem.