HOW TO - Flowroute Trunk with Proper Use of IP Auth and new PoPs


(United States) #21

I have an issue where I can’t get inbound calls to work. out bound work just fine.

looks like part D is the issue. with no route it works. with ip:5060 the call route doesn’t work. I also have a static ip.

###UPDATE###
so with firewall on and set to internet (default firewall) 5060 was not working
I added the IP’s from flowroute and have inbound calls working now.


(John Lind) #22

Problem Resolved - see Edit #2 below

Resurrecting this thread a year later . . .
I’m having massive problems trying to get a FreePBX pjSIP working with one of Flowroute’s new POP subdomains. Works perfectly with the legacy domain in Nevada, using a chan-SIP without requiring any port forwarding in my router, or allowing anonymous or guest SIP calls. I’m not using the FreePBX firewall (module isn’t installed). If I switch over to one of the new POP subdomains on Flowroute’s configuration page, and switch trunks to a pjSIP set up per Flowroute’s instructions, which parallel these for SIP registration, I can make outbound calls with no problem, but get near zero inbound. Perhaps one in 30 (or less) inbound calls works. I’ve been making a huge number of calls from a PSTN line and cell phone to troubleshoot this. Almost always gets nothing. Dead air. Times out after 30 seconds and quits with a hangup (calling end, not the PBX). A look at the logs shows an enormous number of “unsupported transport” errors. If I switch back to the SIP trunk and switch back to the legacy (Nevada) domain, everything magically works again and those errors disappear. Has anyone else encountered this? Anyone have a clue about what’s broken and how to fix it?

Edit:
Got the transport errors to go away - in the Asterisk SIP settings under the PJSIP tab - setting “Allow Transports to Reload” to “No”. Still no joy in Mudville. If I call out to my cell phone and then immediately call back in, it works, but only once. A second inbound and all subsequent inbound fail.

Edit #2:
It’s fixed - or appears to be. Incoming ring as they should. In the process I set up a Dynamic DNS for my Internet connection and set Flowroute to use routes with the Dynamic DNS URL with port number (5060 for pjSIP). Set parameters for that accordingly in the Asterisk SIP settings. I believe that improves reliability over using registration for inbound. Turned off FAX detection (but I don’t think this is what fixed it). Completely started over on the Chan_pjSIP trunk for one of the North America POPs and set the routes up in Flowroute to also use that same “edge strategy” in the inbound route using port 5060. When I enabled everything it started working properly. Something was wrong, perhaps with the trunk or some other setting, but if it’s not broken now I’m not going to continue trying to fix it. AFAIK, the method given for setting up the Chan_pjSIP trunk is good - as a review of my trunk and its setup - with the only deviation being using registration vs IP verification on outbound (also as outlined) doesn’t find any difference. Port forwarding across a router gateway of ports 5060 and 5160 to the internal (static) LAN IP address of the PBX is required when using the Flowroute routes sent to the WAN IP address for inbound (in my case using a Dynamic DNS URL) in lieu of SIP registration…

Thanks
John


(JP) #23

You are incredible for doing this. Any way I can buy you a beer?

A problem I’ve run into that may help others:

In my Flowroute route config, when I set “Inbound edge strategy” to “Default edge strategy” inbound calls just absolutely will not work. Yes, I have set the default strategy in the registration tab; even deleted/recreated the route after setting the default strategy setting. If I set the Flowroute route to the same default I have in the registration tab it magically works.

Replicated on several Flowroute accounts and FPBX systems at this point, all with the same result. No idea why, but it works, and I can’t see any downside. Maybe add this to OP? Or not; I’m not your mother.


(Lucas Ryan) #24

Using this guide, I setup a PJSIP Flowroute trunk and all was well. I did however (for some reason which I now forget) have the Direct Media option set to yes in the trunk. Everything was working perfectly for about a year. Then this weekend, I could get calls, but there was no audio either way. Looking at the logs and console during the live call, I noticed it said something similar to “direct media enabled.” So I went to the trunk, disabled Direct media, and then everything was well again.

So my question is: What is the proper setting for Direct media on a PJSIP trunk, when used with Flowroute? Also, can someone explain the reasoning/details behind the correct answer so I can understand it?


(Dave Burgess) #25

In the large, Direct Media should be disabled unless there’s a really good reason to enable it. With it enabled, the audio is not handled by the server and goes point-to-point from the endpoints. When disabled, this is the feature that allows things like call recordings to work. It also gets around a lot of “double-natted” audio problems, since both legs of the call come through the PBX instead of relying on open firewalls at both ends.


(Lucas Ryan) #26

Thanks Dave. So Direct Media will have no impact on Flowroute and their Hyper Network, failover systems, etc? It might be a dumb question, but their hyper network/failover systems sometimes are a bit of a mystery to me in regards to how they work exactly.


(Dave Burgess) #27

I would expect not - in fact, it might make the whole thing work better, since each session is atomic to a specific ITSP server, so using the FreePBX server as your RTP aggregator is probably a good idea when it comes to your phones anyway.