[Freeswitch-users] FS SIP audio quality?

[jay binks]

Back in November, Brian ( BKW ) was raising money to get new sounds recorded
...
intending to have them for the 1.0.2 release..

I wonder if they made it in, or if they are still coming ...

Jay


On Tue, Feb 17, 2009 at 8:19 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> There is also another side to make mimd to: the Asterisk sounds you
> hear more often (the demo ones) are very long ones.
>
> The ones of the FS demo are very very short (many times just one word)
> and concatenated with the insertion of sleeps.
>
> That is probably someway altering the equation between user experiences
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton
> <nik.middleton at noblesolutions.co.uk> wrote:
> > For what it's worth, using Asterisk recordings, I found FS to be better
> > than when played on an Asterisk system.
> >
> > I came to the same conclusion early on that the included prompts with FS
> > were of a relatively poor nature.  Not volunteering to record new ones,
> > but they do let the product down, as they lead to discussions as below.
> >
> >
> > Regards,
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul
> > D.
> > Sent: 17 February 2009 03:33
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] FS SIP audio quality?
> >
> > I re-tested calls to VM replacing some of FS prompts with * ones, and it
> >
> > appears that * sounds were recorded with a better quality/higher volume,
> >
> > so FS itself has nothing to do with that. That's solved. :-)
> > I am going to double check all  the equipment we used for tests, like
> > headphones, telephone sets, cables since I am almost convinced that
> > there's nothing in FS which can produce effects I observe.
> > I will post back if I find anything wrong, appreciate everybody's help
> > with this.
> >
> > Brian West wrote:
> >> I'm not able to reproduce this issue.. can you verify the codecs are
> >> what you think they are on both Asterisk and FreeSWITCH.
> >>
> >> /b
> >>
> >> On Feb 15, 2009, at 8:04 PM, Paul D. wrote:
> >>
> >>
> >>> Well, I tried several call scenarios:
> >>> 1. Call from X-Lite or Linksys to VM.
> >>> 2. Call from X-Lite or Linksys to a conference.
> >>> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and
> >>> CallWithUs.
> >>>
> >>> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise
> >>> grade Intel server. So just comparing audio in the call scenarios
> >>> above
> >>> * somehow does noticeably better job, sounds clearer and volume is at
> >>> the right level. I am not changing any phone settings of course when
> >>> switching between * and FS.
> >>> I am not biased towards FS or * at the moment, though FS seems to
> >>> have a
> >>> better designed configuration options and community.
> >>> Just wanted to share my experience, and hear some opinions.
> >>> Unfortunately I cannot spend whole amount of time investigating this
> >>> case now, capturing packets etc., but I will try to do that once I
> >>> have
> >>> time. Meanwhile I will have to stick to * for prod.
> >>>
> >>
> >>
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> >>
> >>
> >
> >
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-- 
Sincerely

Jay
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