VoIP Mailing List Archives

[viljoens at verishare....]

Hi Guys

I've got a strange problem - on my asterisk instance, when a call starts to
ring, I do

core show channel <channelname>

and I get the usual output with the duration and billsec fields included.

For most of my calls, things are normal, e. g. duration field starts
incrementing as the SIP phone rings, and the moment it is answered / the
call goes offhook the duration timer continues running, and the billsec
timer starts up. Disposition goes from NO ANSWER to ANSWERED the moment
billsec starts incrementing.

However, for certain calls from a certain SIP trunk provided by a local
trunk provider, this never happens.

E. g. the call comes in on this "problem trunk" and duration timer starts
running - RTP starts and the call is totally normal, both parties have
crystal clear bi-directional audio and the call records correctly - but the
billsec timer never starts incrementing and forever remains at 0.
Disposition forever remains at NO ANSWER - even though the call is in
progress and has been answered, and is working perfectly.

Other calls from other trunks provided by the same provider on the same
logical and physical Asterisk instance work correctly - if the call is
answered, it becomes ANSWERED in "core show channel" display, and the
billsec timer starts incrementing.

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?

Thanks!

Stefan



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[jcolp at digium.com]

Stefan Viljoen wrote:

<snip>