I’m new to VoIP, setting up my first system for my small business. Transitioning off of a traditional POTS system. While I am new to VoIP in general and FreePBX in particular, I am a software engineer and feel pretty at-home learning all the configurations, tools etc for everything.
I think I have a pretty good handle on how things should be set up. Here is where I am in the process:
Purchased a bunch of Clearly IP phones for the office. They are on people’s desks next to old pots phones. Set up FreePBX on a dedicated server also purchased from ClearlyIP, following the YouTube guides from Crosstalk. All the phones on desks are working fine with FreePBX - we can call each other’s extensions, leave voice mails, conference calls, park calls, etc.
I also set up a trunk with voip.ms, again following the guides from Crosstalk as well as the wiki on voip.ms. MOST of the time everything works just great. We can make outbound calls. We can receive inbound calls. I’ve set up ring groups. I’ve got on-demand call recording set up that emails to the owner of the extension as soon as the call terminates. Everything is more or less perfect, and we are ready for our main phone number to port… EXCEPT: seemingly randomly, multiple times a day, I’ll pick up my new VoIP phone to place a test call to my cell phone and I’ll get the “That number has not yet been registered” message. If I also immediately use my cell phone to call one of the DIDs I have at voip.ms, I’ll get a “call failed” message on my phone after it tries for like 30 seconds.
If I simply wait 30 seconds to a minute, and try either an incoming or outgoing call again, everything works fine.
Seems to me like an issue with voip.ms, so I reached out to their support, and explained everything. They insist that the problem is with the INVITE format with my SIP sessions, and that I have to follow the following format:
From: "CallerID name" callerID number < sip:[sipaccount]@[server].voip.ms > To: < sip:[Number]@[server].voip.ms > Contact: < sip:[sipaccount]@[your_IP]:SIPport;transport=UDP >
In using sngrep to inspect my SIP traffic, I see though that on successful calls that format is never used, and instead there is a challenge/response sort of back and forth where FreePBX initially sends an INVITE which is met with a 401 UNAUTHORIZED, which FreePBX responds with ACK, then a fresh INVITE that includes and Authorization: header. The response from voip.ms is then 100 Trying, then the session starts up.
My FROM: format looks like this:
From: "COMPANY NAME" <sip:[my cid redacted]@[ip redacted]>;tag=[uuid redacted]
I have not yet used sngrep to inspect a failed call (just learned about sngrep this am, and have been periodically trying calls all am - but of course it has not failed yet).
I assume that my registration with voip.ms is dropping, but when a call fails the voip.ms user portal says my FreePBX box is currently registered (there could just be lag there, though - it’s not super-real-time), and if I grab trunk status from asterisk it always shows something like:
Endpoint: VOIP.ms Not in use 0 of inf OutAuth: VOIP.ms/[username redacted] Aor: VOIP.ms 0 Contact: VOIP.ms/sip:[username redacted]@seattle3.voip.m d56ce2f3c4 Avail 13.893 Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5060 Identify: VOIP.ms/VOIP.ms Match: 18.104.22.168/32
So I have a few questions:
- Is voip.ms support nuts? Are they trying to get me to configure things for the older chan/SIP, rather than PJSIP?
- I’ve seen several older forum posts where people report exactly the same behavior, and simply changing to a different endpoint at voip.ms seems to have solved the problem… but did it really? Or did those forum topics just auto-close when people gave up trying to solve the actual problem? I really like that 13ms ping time to the current endpoint, and would like to stay geographically close to me.
- Does anyone here have good experience with voip.ms? Should I be looking for a different trunk provider? Their support has been… less than stellar. I’ve been back and forth with them more than a dozen times since Thursday, (nearly a week now) and they insist that FreePBX is only for advanced users and they don’t really offer “advanced support”. trigger eye roll