Hello!
I am using an onsite FreePBX deployment with a cable company ISP. While, it isn’t unusual to have this problem, we have definitely had more trouble since our illustrious cable company “upgraded” their hardware (infrastructure, not their modem in our building).
Here is the situation: We dial a cell phone (I have tried it on mine even), and get a busy signal (despite the cell phone not being busy). You can see the log here: https://pastebin.freepbx.org/view/1ea275be
It is not possible to replicate the problem - my own phone was beside me -not being used. It did not ring and was reported busy 1 out of 4 tries. (and rang the other three).
When the outbound call is to an actual landline, it works every time.
During the upgrade, I started setting up a FreePBX VPS (Hosted by CyberLynk). I stalled out when it was time to connect the phones to the server (we currently use DHCP Option 066) and by the time I had time to review the instructions, the upgrade was finished. Do you think the VPS would help us out? Is there anything else we could try?
What is 192.159.66.3? Your problem lies there. Is it where you intended to send the call? If not, is it your router, in which case, the first step is to disable SIP ALG, in it, by whatever name it is called.
Note that there is no real way that Asterisk would know the destination is a mobile
Thank you, @david55 for your response. I am running a UniFi Dream Machine Router - SIP ALG has been disabled. It’s internal address is 192.168.10.1. I looked up the 192.159.66.3 address and found that it owned by Wisconsin CyberLynk Network. Is that our SipStation server?
Probably; you should know what domain you are sending to, and can look up the address. Basically that is sending the Busy Here. It’s either misusing it, or it believes the final destination has reported busy. It’s not something that it should send for any error in Asterisk, other than than sending a phone number that is wrong, but addresses a valid, busy phone.
That would be trunk1.freepbx.com, the primary SIP server for SIPStation. If you contact Sangoma technical support they have the ability to go look at the full SIP signalling across things and see what the upstream did and go from there.
Thank you @jcolp & @david55
To be clear, so far, it doesn’t look like there is something wrong with my setup. I will contact SIPStation to see if they can shed some light on it. Do you mean the “busy here” signal could be being misused?
I wouldn’t use the word misused, but something somewhere thought it was busy for some reason. Determining why would need to be done, and that is upstream so out of your direct hands.
When I said misused, I was contemplating that it might be translating some other condition than the phone being busy, e.g. an inability to find a free radio channel, as being subscriber busy.
You have quite a long chain of different signalling protocols, and it is possible that busy and congested cases are getting mixed up somewhere on the way, but Busy Here should only really be used for real cases of busy.
I am setting up a SIPStation support request now. Thank you both! I get an abnormally high number if Fail2Ban notices after I post logs to PasteBin. I suppose that’s to be expected…
The cell phones you are having problems with, have you narrowed it down to any particular provider ? We are in Hawaii and are having problems with 1 way audio. But it is only on calls that interact with a T Mobile account. So far we have tested this against Verizon, H2o, AT&T and a few others, but so far 100% of the failed calls are T-Mobile. We also have an open ticket with Sangoma about this.