Calls drop at exactly 29.5 minutes using provided ATA

Ok this morning at 11:02 I was on an incoming call from Boston to my 212 voipo number. I had simul-ring turned on. I was speaking to the person successfully on my cell (the only target simulring number) and bingo! at the 30 minute mark, the call disconnected so it cannot be a function of the terminal adapter since the call was not connected through the adapter.
Should I file a ticket or not?

To file or not file....that is the question.
Whether 'tis nobler in the mind to suffer
The slings and arrows of outrageous cutoffs,
Or to take arms against a sea of calls,
And by submitting a ticket, end them?

Oh my. It sounds almost like the server-side code that is doing the RFC4028 stuff (session timers) is still running, and tears down the connection when no reply comes back. If it is simul-ringing your voipo adapter and the cellphone, it's possible (maybe?) that the voipo server starts the session timer. When your ata doesn't answer, but your cell does, one would think the tiimer would be cancelled. Or maybe it is erroneously starting the timer in the first place. Sounds like a ticket is definitely in order...

Maybe Voipo will do away with the rfc4028 timer and everyone will be happy ;)

They must have done something because my calls do not cut off anymore at 29.5 using the ATA. I dunno about the other.

I'll try later today on my BYOD and see if anything has changed there.

My calls are now dropping at 11 min and 15 sec. I submitted a ticket.

Quote:

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Originally Posted by Montano

Maybe Voipo will do away with the rfc4028 timer and everyone will be happy ;)

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I can't remember, but what is the benefit to the timer? Does it clean up broken registrations or something?

Quote:

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Originally Posted by scott2020

I can't remember, but what is the benefit to the timer? Does it clean up broken registrations or something?

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This is a bit dated, but may explain its purpose..

http://rfc.sunsite.dk/rfc/rfc4028.html

Something must surely have changed since I just placed a 32:41 minute call through the * 1.4 box and it was uninterrupted.

The bad news is I have mysterious [jerks(?)] from 93.190.137.247 and 85.25.85.70 repeatedly coming into the * box via SIP for whatever purpose... guess they just have to try my default "s" destination out! :D Just in case anyone else here worries about or watches for such events!

Is anyone still seeing issues with calls dropping?