[asterisk-users] 1.8.32.3 - billsec field does not not increment after call answer - what triggers it? (Joshua Colp)

[Stefan Viljoen]

Hi Joshua

Thanks for the response.

Interesting that you mention that toll-free numbers can do this, this
problem trunk happens to receive calls from the national telecoms provider
here (Telkom SA) sourced from a toll-free number. The SIP trunk provider has
ported that toll free state telecoms company number for us to a local trunk
number with which we authenticate as a SIP peer. (Not sure if what Telkom
here in South Africa define as "toll free" on a technical level match what
you mention as "toll free" on an American / United States model.)

The thing is the number stays in the unanswered state (while in fact
answered and working perfectly) for the entire duration of the call, so then
it appears 200 OK is not received for the invite.

I therefore assume that the actual RTP negotiation process does NOT need the
200 OK for the invite, for the call itself is fine - it is just that
Asterisk never "realises", as regards the CDR, that the call was in fact
answered.

At least it is working and audio flows back and forth.

I'll see if I can come up with a SIP trace.

Thank you!

---
Stefan Viljoen wrote:

<snip>

>
> Only this one trunk consistenly has this problem for all calls received
over
> it. The trunk provider is using sippy on their side.
>
> What setting / config option for the particular SIP "problem trunk" have
my
> trunk provider changed on their side to stop Asterisk from recognising
that
> a call has been answered when it comes in over that trunk?
>
> It appears some SIP traffic is not being sent by them (or not received by
my
> Asterisk) that indicates to it a call has been ANSWERED and that it must
> start the billsec timer?

I can't really speak for the provider but some numbers will stay in 
inband progress (unanswered) for a bit. Some toll-frees for example.

The specific SIP message that would show it as answered would be a 200 
OK to the INVITE we sent though. If you provided the SIP log then we 
could see.

Cheers,

-- 
Joshua Colp