You Anyway to estimate how much ram needed for 30 calls a minute?

[Ben Uecker]

radio station does a birthday call in. It last for 10 minutes for 6 days a week. The first few calls get through then the lines have. I one on the other end for a few seconds and then less than a minute later the calls start coming in again. I checked the call log and it looks like it’s about 30 calls a minute.

The ram looked like it hadn’t maxed and I will need to go back and check the log again instead of in real-time but I think it’s a ram issue. Too many calls at once. I have six gigs assigned.

 

[tbrummell]

radio station does a birthday call in. It last for 10 minutes for 6 days a week. The first few calls get through then the lines have.

Have what?

I one on the other end for a few seconds and then less than a minute later the calls start coming in again. I checked the call log and it looks like it’s about 30 calls a minute.

Huh?

The ram looked like it hadn’t maxed and I will need to go back and check the log again instead of in real-time but I think it’s a ram issue. Too many calls at once. I have six gigs assigned.

Your post isn't making much sense. maybe you could re-post.

Things I'd post:
Inbound source (SIP, PRI, Analog). If not SIP, gateway brand and channel config. If SIP, SIP provider and purchased config (IE: max channels)

Inbound routing: where do you send the calls

Destination endpoint type: What do these calls get answered by and what is the integration? Some type of broadcast console that does SIP? Broadcast console with POTS (meaning there must be a few ATA's in use)?

Relevant call logs of a failed call.
The more info the better if you want any relevant help. Otherwise it's all speculation.

 

[Ben Uecker]

Sorry, my English skills take a hit when I am working all day with West Indians.


Call in: as in people call in to the radio station.

No one on the other end: as in you pick up a ringing line and you only get dial tone.

inbound routing: call forwarding enabled at the telephone company. That is forwarded to a voip.ms number. That is sent to an on-prem wazo setup IVR. Pressing "1" rings two phones. Studio B and Studio A.

The console does not play any role in the phone system. They simply put the mic close to the phone.
Internet is provided via an ONT connection on fiber.

In terms of max channels, voip.ms shares them with all the sub accounts. I think I am up to 25 channels for my clients. I have considered lowering the max channels setting for the user in the pbx.

I don't have the time to try and pull any logs today.

 

[tbrummell]

OK, so the way I see it:
1. PSTN number via XYZ Telco, call forwarded always to Voip.ms DID
2. Voip.ms DID directed to Wazo PBX, answered at IVR
3. Caller presses 1 and is routed to 2 phones in a Ring Group? Ring all I assume?
4. Caller answered by studio personality

Assumptions:
Ring Group is ring all
2 phones are SIP devices
Voip.ms does not limit inbound max calls on a per minute plan as far as I am aware
Voip.ms does limit max calls on outbound per minute plan (must open a support ticket to increase, but not relevant to this trouble)

1 more question about the SIP endpoints: how many calls can they handle at 1 time when in the ring group scenario? This is probably the source of your issue. This is of course assuming you are using SIP devices. If using POTS/ATA, well, good luck.

If they are SIP devices and the device supports multiple accounts, I would add multiple accounts to the phone(s) and then add all of those accounts to the ring group. You'd still need to match the number of concurrent calls that the phones can handle to what Voip.ms is sending you.

*OR*

Setup ACD. Log the 2 phones in. 2 callers are answered, other callers are in queue.

I don't use Wazo, so I don't know what it is capable of.

A SIP trace of a failed call would go a long way in troubleshooting. My guess is both phones send a 486 Busy once it's handling/ringing with the max amount of calls it can handle. Hence why the caller ends up in never-never land.

 

[Ben Uecker]

Yes to all. The ring group was first tried as ring all then for this morning I disabled the setting "call a member already in line." These SIP phones are Yealink T46G. I have 5 lines setup.

Yes, I was thinking maybe an ACD would make a difference. I have to figure that part out.

Now I have some time and am going to try and get the logs from this morning.

 

[billsimon]

I agree that setting up a queue is the most logical on-prem solution for this.

At the same time this kind of thing makes me think providers like Twilio, where you can build applications on the platform, might be useful. Save your Wazo for your PBX and use Twilio (or newcomer Signalwire) for applications that receive and process these bursts of calls. Just an idea.

 

[Ben Uecker]

I have been playing around with Twilio for as long as they have offered trunks but have never taken it into production.

I just saw signalwire last night on the site. It's also something that I will be testing out in the coming months.