Calls Die on Hold Pt. 2

Toronto, Canada

Add me to LinkedIN

LXLS · Jan 25, 2021

Curly, Good to know. I used the term "die" simply because the call can still be seen as active in BCM Monitor when this happens. The call just goes silent but never actually drops. Now I know!

I do have the IP entered there, yes. Due to being on a dynamic IP, STUN is turned on, with the default stun.ekiga.net -- It figures out the public IP and populates the field.

The Media address according to the carrier should be my public IP all the time, but when put on hold, it changes to 192.168.1.90, which is the IP of the BCM on my local network.

allworxguy · Jan 25, 2021

What is confusing to me, and i might out in left field, why should the ITSP care or even get an indication that the call is on hold. That's an internal function of the BCM. The BCM shouldn't even be broadcasting a state change to the ITSP.

What about "Tones" on hold? What were using for a music source?

Marv ccna

curlycord · Jan 25, 2021

I do not actually use stun any longer, I have 5 different SIP carriers and found it to be a hinderance where some sip trunks were effected.
Try disabling it an manually enter the address.

Check this thread, maybe something in there might help.

Are you absolutely sure every little setting in every tab of SIP Trunking is the same as the other BCM?

Are you able to to put the BCM on an older router and direct to your mode (avoid current network) to test?
Or bring the BCM to the other site?

What about using only G729 as a codec in Media Parameters?

Outta Ammo...again!

________________________________________

=----(((((((((()----=

Toronto, Canada

Add me to LinkedIN

exsmogger · Jan 25, 2021

I've used Flowroute on my BCM50 R6 for many years with almost no problems. Post your SIP settings and I'll see if anything differs in my setup.

Brian Cox
Georgia Telephone

LXLS · Jan 27, 2021

Allworxguy:

From my understanding, this is normal behavior for SIP across most platforms. In this case, the BCM does a blind-transfer to BcmAmp which kicks off a re-invite to change the RTP stream's destination. (again, from my understanding.)
Tones has the same issue, I've been using the onboard Music Manager which as far as I'm aware is the service "BcmAmp"

Curlycord:
I disabled STUN and entered the IP manually, no change.
I scoured your sourced thread and didn't find anything that tipped me off.
I put the BCM on an older router, and even exposed the device directly to the internet with zero NAT or Firewall, no change.
G.729 only, also no change!

exsmogger:
here's a zip file with a bunch of screenshots showing how I've got FlowRoute set-up on my system.

exsmogger · Jan 27, 2021

LXLS,

I went through your screenshots and there are many differences from my settings. I'm attaching the screenshots from my own system where there are conflicts with your settings.

Brian Cox
Georgia Telephone

LXLS · Jan 27, 2021

Brian,

Thanks for doing that! There are indeed quite a few differences. I mirrored your SIP settings, changed every option one at a time and the issue remains.

For now I'll use the silence option for MOH, I think I'm going to ask around and see if anyone has a blank, unconfigured R6 image that I can use to test with because I have now wasted hours of my life trying to sort this issue out with what I've got, and I'm beginning to think something may have really gone wrong on the back-end.

curlycord · Jan 27, 2021

You SIP programming differs so much from my SIP accounts and Dest/Route/Plan programming that I am just going to shut my mouth and wonder how yours even works at all, maybe I play later someday....lol
That's the thing with BCM, so powerful it works in different ways.

Will let exsmogger work with this one since he use Flowroute.

________________________________________

=----(((((((((()----=

https://www.tek-tips.com/faqs.cfm?fid=7741

Toronto, Canada

Add me to LinkedIN

LXLS · Jan 27, 2021

Well, I'm not a telco guy, but a sysadmin. Phones are definitely not my specialty, I had to piece this together by reading the manual and as many forum posts as I could. Have I gone about setting up my Dest/Route/Plan programming up in the worst possible way? what exactly, if you don't mind me asking do you see wrong with the way I've set it up? Genuinely, this question is an attempt to gain more knowledge on something I'm not trained specifically on.

curlycord · Jan 27, 2021

If it works it's not necessarily wrong.

The one thing that stood out was that in SIP Trunks/RoutingTable you have 1 as the Destination Digit
The 1 would then match the 1 in the DialingPlan/Dest Code, but you do not have a 1 in your Dest Code programming, nor the Public Network/DN Lenghts.

So it seems I need to figure out how it works that way.

Here is my FAQ's that might indicate how I have programmed my lines:

Perhaps I can add Flowroute to the FAQ.

Found this but for CISCO:
Support: try disabling Supports Re-invite
User: That worked!

________________________________________

=----(((((((((()----=

Toronto, Canada

Add me to LinkedIN

exsmogger · Jan 27, 2021

I have 2 guesses on the original problem. "Enable local NAT compensation" under SIP Trunking-Public-Advanced should not be checked. That will definitely hose the voice ports navigating the local router's NAT and cause the lost calls on hold.

In the same area under ITSP association method you should select "From header proxy address match." This is what a Flowroute tech support gent told me it should be set to.

Brian Cox
Georgia Telephone

exsmogger · Jan 27, 2021

Another thing is to definitely disable the "Session refresh method." The setting above it for "NAT pinhole maintenance" serves to keep the connection alive. I spent many hours tracing packets through the BCM50 and found that the "NAT pinhole maintenance" setting was the key to keeping Flowroute working smoothly.

Brian Cox
Georgia Telephone