I’ve set up a conference line with its own DID and Extension. This is FreePBX Conference. Not pro. The room works as expected until the 3rd or 4th caller tries to join. They reported being unable to connect and the call is ended. I’ve verified that my conference config is set for up to 10 callers. The only thing I could come up with was that maybe this setup would require a high volume trunk, but I’ve been unable to verify that anywhere. Any help greatly appreciated!
This is likely a channel limitation imposed by your trunking provider. If this is the case, the rejected caller will usually hear a busy signal or error announcement, and the Asterisk log will show nothing at all for the attempt. However, internal callers dialing the extension for the conference room (not the DID) will still be able to connect.
If this is not your situation (failed calls do get logged), please post the Asterisk log for a failing attempt.
SIP trunks are usually priced in one of three ways:
Low monthly cost for number, but you pay a per-minute charge for incoming calls. In this case, the channel limitation is only to protect against attacks and bugs causing accidental loops. Generally, you can open a ticket with the provider and ask them to increase the channel limit.
Low monthly cost for number, but you pay a significant monthly fee for each channel. Channels are shared across all numbers in your account. There is no per-call or per-minute charge for incoming calls. In this case, you would either need to buy more channels, or get a DID with a different plan.
Higher monthly cost for a number that comes with a small number of channels, usually 2 to 4. There is no charge for incoming calls. These DIDs are intended for homes, very small businesses, or personal numbers for a larger business and are not suitable for your application.
You may also consider using a conferencing service instead of hosting your own.
For more specific advice, please post:
What country is the DID in? Who is your trunking provider? How often would you be having these meetings? How long do they typically last?
Yep. You were on the right track. I contacted SIPStation. I either needed to add more trunks or enable concurency bursting. Thank you!
Unless your meetings are really just lectures (everyone but the presenter is muted), IMO FreePBX doesn’t work well for conferences with more than about six people. Take a look at Communicate their desire to ask for some commercial alternatives.
If you want to host it internally, consider getting a supplementary trunking provider. In addition to cost savings, meetings wouldn’t ‘tie up’ your SIPStation trunks, leaving the lines open for customers calling in.
One that would allow you to test right away is https://signalwire.com/ . They give you a small credit at signup, sufficient to buy a DID and have about two hours of a 10-participant conference.
Unfortunately, they are not (yet) a real trunking provider – the service lacks failover, 911, international, decent logs and debugging tools. However, you may find it adequate for this application. (Set it up on chan_sip; I had issues with pjsip that I didn’t take the time to investigate.)
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