Inbound Call Floods get Error instead of Busy

We recently made the switch from analog POTS lines with hunt groups (1+3 rollovers) to VoIP. We have two redundant SIP trunks with 25 session channels to an Asterisk/FreePBX server version hosted in house. Throughout most of the day, our services work as expected. Inbound calls are correctly routed to their assigned destination chan_pjsip extensions with a Follow-Me configuration acting as a rollover for three additional extensions. When all four are in use, a busy signal is returned to surplus callers. Inbound Routes, Extension configurations haven’t deviated much from system defaults.

During very high volumes of incoming calls (short duration contests), instead of getting a busy signal some incoming callers are getting a “due to technical difficulties your call cannot go through” message. These same callers are able to ring through outside of these incoming call floods, but they’re sometimes hearing these messages during the incoming call flood, sometimes not. Our CDR’s indicate up to 250+ incoming calls a minute during these incoming floods.

Response from SIP Trunk provider:
Conflicting responses from our SIP provider’s tech support. One person I spoke to on the phone (who submitted a ticket internally, providing no reference number) claims the incoming caller’s carrier will return the “…call cannot go through…” message any time our 25 session channels are exceeded. When I submitted my own support ticket to our SIP trunk provider with sample call details a different tech emailed back stating that “all calls rejected due to capacity limits will return a standard busy signal”, attaching a Call Flow showing a “603 Decline”, confirming our PBX was receiving and rejecting the incoming call.

What We Need to Figure Out (any insights appreciated!):
Clearly we’re exceeding an inbound threshold during these contests, but where? Our SIP Trunk provider’s or a FreePBX configuration? We need all surplus callers to hear a busy signal, never a “…call cannot go through…” recording. Given the “603 Decline”, I’m focusing first on our FreePBX configuration.

  1. Does anyone know where the “Due to technical difficulties, your call cannot go through” message would likely be originating from? Our SIP Trunk provider? Inbound caller’s carrier? Our Asterisk/FreePBX server? The “603 Decline” suggests it might be the latter. Is there a list and discription of error message recordings for Asterisk/FreePBX? I haven’t yet been able to locate one.

  2. Is there a FreePBX configuration setting where I could effectively send a busy signal to any inbound call when all four destination extensions (1 + 3 rollovers) are ringning or in use? I’ve tried changing the Device State Busy At value in Extension# > Advanced > Edit Extension but that didn’t seem to make force a busy signal after the value entered.

Basically your PBX can handle 25 concurrent calls and either reject the call or answer it and send it wherever including playing a message or whatever, any calls past those 25 will be handled by your carrier and are thus out of your direct control. Perhaps they provide you an alternative ‘overflow’ destination , perhaps they don’t .You will need to sort that out with them though.

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Thanks dicko. Are you saying FreePBX itself has a max cap of 25 concurrent calls? Can I assume this includes ringing / not answered calls in addition to those in use / on hold?

No, I am saying that if you pay for 25 concurrent calls, don’t expect the 26th one to be sent to you. That applies to all calls to your DID, you get to handle 25, ringing or up. The 26th+ you will never see.

Understood. I just want the 26th to hear a busy, not an error message. Many callers are getting that busy signal but we seem to pass a threshold during floods where some are getting error messages. So you’re agreeing with the Trunk provider’s first support person’s response - that beyond 25 should result in an error message… I’m still left wondering why the second support person was able to show me a 603 Decline in our Call Flow. That implicates our FreePBX/Asterisk system.

Only your carrier can provide that facility or indeed confirm their tech support. If they sent you a 26th call then the second support person is correct, if not then the first one is.

You can independently watch arriving calls with sngrep

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25 calls is quite a light load.

Nonetheless, if you really are returning 603, there is not a lot we can do without seeing the logs from Asterisk’s side.

Agreed about the logs, @TueNiteLive would need to confirm that the 603 was sent by his PBX in the obfuscated picture in his original post and not possibly the response he got from his carrier for a coincident outbound call on the same overcapacity trunk.

I may have found something relevant. Combing over my FreePBX configurations, I found (and have now deleted) a Ring Group entry for the same set of destination extensions I’ve been getting these complaints about.

This Ring Group is a remnant of an early attempt at a rollover. After initial testing at the time, I later decided to go with a Follow-Me configuration to achieve rollover. I’m hoping that deleting this Ring Group from earlier might put an end to the “…call cannot go through…” recordings and inbound callers will hear ringback or busy instead. I’ll follow up tomorrow.

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