We’re using Asterisk 184.108.40.206 with FreePBX (started as AsteriskNOW but updated Asterisk).
The problem I’m having right now is that when I try to originate a call from the meetme conference channel, I don’t hear any ring or busy as the call is dialing.
The originate uses channel of Local/xxxxxxx@meetmehere where xxxxxx is the meetme room # and the meetmehere conference just calls the meetme application with the exten as room #.
What we’re trying to accomplish is:
- Receive call to SIP station (from SIP trunk)
- Place inbound trunk channel on hold by redirecting to MOH context
- Dial external PSTN # from the SIP station
- Speak with the dialed party from the SIP station
- Patch the two calls together (inbound and outbound)- dropping the SIP station
I’d like to be able to accomplish the above with the manager interface handling everything with “HOLD” button and “DIAL” button. All the SIP station operator has to do from the phone is answer the call.
What we’re currently doing that almost works perfectly is:
- Recieve call to SIP station from SIP trunk
when the hold button is pressed:
- Redirect both channels of call via Redirect (using channel and extrachannel) to a context that plays MOH
when the dial button is pressed:
3a) Redirect the SIP station channel to a meetme conference
3b) Originate a call, dialing the external number from the meetme conference
when the transfer button is pressed
- move the original inbound channel and the final outbound channel to a different meetme room and drop the agent
Another question might be… Is there a better way to accomplish what we’re trying to accomplish?