Hello, I have a freepbx 13 on asterisk 13.
What we are trying to accomplish is to have any inbound calls that exceed the call limit to rollover and play a busy tone. Currently they go back to our sip provider as a failed call and they forward calls to a roll over number (typically our cell phone).
Do you mean after a specific time elapses, you want them to immediately hear busy tone then get disconnected?
Partially. This DID is getting hammered with an abnormal amount of calls and our trunk is only set up to allow 6 calls, after that it gets thrown back to our sip provider as a failed call which then rolls over to a backup number. What we want to accomplish is to direct the call instead of back to the sip provider to roll over to a secondary “fake” trunk that will just play a tone (fast busy, etc.) and hang up.
If your Trunk is busy then there’s no way how the PBX can receive the call and play a busy tone.
The only other way i can think of to accomplish this would be to get another Trunk and DID from your provider to your PBX, setup the new inbound route for this DID to point to a busy tone, have them route the failed calls to the secondary Trunk.
My question is, do you still want the backup number that your SIP provider has to be used in cases where its actually needed? (i assume you intend this number in the event of a total sip failure?)
The situation is that in order for your PBX to play congestion message to your callers, it requires the use of a trunk.
What is potentially possible (taking your trunk capacity into consideration) is to allow 5 calls in, and then the 6th call just play the congestion message for a couple of seconds then issue a hangup. Depending on how your calls come into your PBX this is easy…or cumbersome to configure. Direct dial extensions vs a Queue or ring group.
This COULD sort of service what you are looking for, but in a situation where you have 5 people on a call, then the 6th person calls, and is hearing congestion, if a 7th (or more) person calls in during the time this congestion message is being played, that call would spill to you other line (assuming no change is made)
Is there a way to create a trunk in FreePBX that just points to a ring group or IVR that plays a sound and terminates the call?
Meaning after the first trunk call limit is exceeded (one we would impose so as not to flood the trunk and cause it to fire back to the sip provider), it would roll over to the second trunk and just play a busy sound.
I am thinking: set up a queue with the capacity>max callers set to 6. Then set failover destination to terminate call, with the congestion option chosen. Only problem, it doesn’t hang up after a few seconds, but relies on the caller to hangup. Would like to find a more suitable regional line-congested tone that hangs up itself. With the set up described, might have to set max callers to 5 so the 6th call into the queue goes to busy. But I guess the 7th caller would still bounce around in the sip providers rollover number. So perhaps does not fix this.
We have unlimited incoming calls, which overwhelms our receptionists, so I am going to try this.
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