[Freeswitch-users] Audio Lost after 1 min

Sat Apr 20 20:09:55 MSD 2013

Anthony - I think you got it backwards.  This is a FS user calling out of FS
and getting voicemail elsewhere, like Sprint.  

In follow up she is "sure" the audio cut off in the middle of the callee's
outgoing message.  Other users at that site with the same phones report the
same problem,

I have called the same number with a cisco phone from a different external
site and do not have the problem.  I have sent are a list of tests to see if
we can reproduce the problem.  Then I will pastebin mountains of data for
you guys.

My current hypothesis is some noise in their office environment caused the
b-led audio on her phone to cut out in favor of a-led audio momentarily.
Then when it returned to b-led audio, the remote end was already past the
"beep" and recording. That could be why it is specific to voicemail calls.
I know some speaker phones can do that, but could Polycom 335 handsets do it
too?

Thank you again for your time and efforts.

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, April 19, 2013 5:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Audio Lost after 1 min

Try setting the channel var record_waste_resources=true in the dp before
calling VM or globally in vars.xml

This will cause useless silent RTP to be fired at the caller while they are
recording.  Without this it logically stops sending while its recording the
message since you don't need to hear anything while you are recording your
message.  Some devices panic with no media and hangup thus this
workaround....

On Fri, Apr 19, 2013 at 11:35 AM, Michael Collins <msc at freeswitch.org>
wrote:

This is interesting. I suspect she has given you only the facts as she sees
them and not necessarily all of the facts.

I believe you'll need pcaps of the calls in question to know for sure.

As far as FreeSWITCH being able to tell the difference between a human and a
voicemail/answering machine... not without a special module. :)

Get those pcaps and FS logs and see if there are any clues. If you can
compare a working call against a non-working call then that might help you
narrow down where to look.

-MC 

On Fri, Apr 19, 2013 at 7:55 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:

Hi All,

I have a report by a customer at a site that we had considerable trouble
with NAT issues.  I thought everything was now stable, but it sounds like it
is not.

She reports that she is only telling me now, but for some time when they
call someone and get no answer and get directed to voicemail (the latest was
a Sprint number) the audio from the remote answering machine cuts off before
then end of the message.  She did point out the call timer on her phone
continued normally.  She swears this does not happen when talking to a live
person!

First, I don't think FS can even tell if the b-leg was answered by voice
mail differently from a person.  Please correct me if that is incorrect.

So, I am proceeding on the assumption that calls that are answered take less
than 60 seconds!  45 seconds of the reported call were waiting for Sprint to
give up on the cell phone and route to voicemail.

The log just shows the call being answered . then Normal Clearing.

I tend to always suspect NAT or crappy Polycom 335s for this customer as
that has been all my issues with them so far. Is there a NAT issues relating
to max call length, etc?  No other users (all cisco phones) on the same FS
at other sites report this issue.

Thanks in advance.

Sean

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_________________________________________________________________________
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