[Asterisk-Users] Confused !

Sun May 14 07:46:01 MST 2006

thanks i alreday did that , you r very helpfull thanks again

/Salaque

On 5/15/06, AR Tarzi <artarzi at batelco.com.bh> wrote:
> 1. In the extension definition, insert canreinvite=yes for each of your
> clients.
> 2. In the trunk definition, insert canreinvite=yes
>
> Read about reinvite at
> http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
> Apparently some hardware does not like it, and obviously, both the client
> and the service provider with have to be able to use the same codec (for
> them to be able to talk to each other) but better if Asterisk is restricted
> to that codec on both sides to start with.
>
> Please understand, I am trying to help and I don't know which parts (of what
> I'm saying) are not entirely accurate but normally if I say something wrong
> there are enough people who clamour to correct me.
>
> ----- Original Message -----
> From: "Mohammad Salaque" <msalaque at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Sunday, May 14, 2006 14:16
> Subject: Re: [Asterisk-Users] Confused !
>
>
> how to use reinvite  in my asterisk setup ?
>
> thanks
> Salaque
>
> On 5/14/06, AR Tarzi <artarzi at batelco.com.bh> wrote:
> > I'm not an authority
> > but why don't you get some g729 codecs (10 or so) and use g729 all around.
> > Not allowing for ADSL overheads you can calculate your own requirements on
> > http://www.asteriskguru.com/tools/bandwidth_calculator.php (please double
> > the results since each call is turned around to your service provider.)
> >
> > I would have thought it would be better if you could use reinvite to let
> > your clients speak directly to your service providers. Someone who knows
> > better ought to be able to tell if this would work.
> >
> > Your restriction is what the service provider allows. Most (that I've
> > used)
> > allow g729. I know it uses more bandwidth than g723 but nothing like G711
> > (ulaw or alaw) and from my experience, the quality is quite reasonable.
> >
> > ----- Original Message -----
> > From: "Mohammad Salaque" <msalaque at gmail.com>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > <asterisk-users at lists.digium.com>
> > Sent: Sunday, May 14, 2006 11:27
> > Subject: Re: [Asterisk-Users] Confused !
> >
> >
> > thanks for your replay,
> >
> > after i disallow all codec except g723 i also confused how a2billing
> > is working then what i did , i removed all the codec from
> > /usr/lib/astersik/module without codec_g723.so .
> >
> > then i saw in my log while user calling to my ivr access number a2b is
> > looking for gms codec as all the audio file is in gsm format. but what
> > my understanding was it should drop the connection as i only allow
> > g723 .
> >
> > what is found today from one of my frnd telling me that actual
> > bandwidth calculation
> >
> > "
> > For codec g723 incoming and g723 outgoing we need: 48.89kbps
> > For codec g723 incoming and g711 outgoing we need: 114.03kbps
> >
> > So, to run 8 calls we will require 902.15 kbps or 0.9 mbps
> > For 10 calls we need 1140.3 kbps or 1.1mbps
> >
> > Each call has RTP, UDP, IP, Codec and SIP overhead.
> > "
> >
> > so what u guys suggest , should i record all my ivr file in g723
> > format all . increase my bandwidth!
> >
> > /Salaque
> >
> >
> > On 5/14/06, AR Tarzi <artarzi at batelco.com.bh> wrote:
> > > Unless "reinviting" works, wouldn't that add up to what he's
> > > experiencing
> > > ?
> > > client <-> asterisk <-> service provider.. makes that 180k each
> > > connection
> > >
> > > so 4 of them would give 800k or so.
> > >
> > > What I can't understand is: if only g723 is allowed, and Asterisk only
> > > allows it as passthrough, how's the A2billing IVR working ? I have to
> > > assume
> > > G711 (ulaw or alaw) is used.
> > >
> > > ----- Original Message -----
> > > From: "Woodoo People .pGa!" <wpeople at Shadow.microsystem.hu>
> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > > <asterisk-users at lists.digium.com>
> > > Sent: Saturday, May 13, 2006 23:36
> > > Subject: Re: [Asterisk-Users] Confused !
> > >
> > >
> > > > Install iptraf, that will allow you to check incoming and outgoing
> > > > traffic
> > > > (or trafshow what do that on /host basis, but not so detailed info)
> > > >
> > > > If you choose ulaw, that should take about 90kbps fullduplex traffic.
> > > >
> > > >> I'd like to share something u all ,  so that i could understand whats
> > > >> going on into my  Asterisk box.
> > > >>
> > > >> i have a setup like this
> > > >>
> > > >>
> > > >> client(ip phone) -----ip network------- [Asterisk]----ip network
> > > >> -------[Service provider]
> > > >>
> > > >> i have configured A2biling in my Asterisk box. so when client call to
> > > >> my Asterisk
> > > >> A2billing's ivr respoce , my client authenticate there pin and call .
> > > >>
> > > >> all my IVR file is gsm format (i got that from a2billing by default)
> > > >> i configured each client
> > > >>
> > > >>
> > > >> disallow=all
> > > >> context=from-internal
> > > >> canreinvite=no
> > > >> callerid=device <20004>
> > > >> allow=g723
> > > >>
> > > >> so client is only using g723 i think..
> > > >>
> > > >> but the problem i am facing now . when there  are 4 calls in my
> > > >> server
> > > >> i saw my bandwidth reach around 1 mbps /1 mbps .  why my server
> > > >> taking
> > > >> so much bandwidth ?
> > > >
> > > > --
> > > > WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com
> > > > wpeople at shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople at RedHat.users
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