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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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 <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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 <[EMAIL PROTECTED]> 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!" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@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
> > [EMAIL PROTECTED]@RedHat.users
> > _______________________________________________
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> >
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