[asterisk-ss7] asterisk oh323 - chan-ss7 echo problem

[leonimar cape]

Hi Goksie,

Right now I am using h323.conf for the quality is much
better compared to oh323. I am also using the A104D
sangoma card and chan_ss7 with the echo cancellation
patch.

h323.conf

[general]
port = 1720
bindaddr = 202.58.251.35        ; this SHALL contain a
single, valid IP address for this machine
tos=lowdelay
;
; You may specify a global default AMA flag for iaxtel
calls.  It must be
; one of 'default', 'omit', 'billing', or
'documentation'.  These flags
; are used in the generation of call detail records.
;
amaflags = default
;
; You may specify a default account for Call Detail
Records in addition
; to specifying on a per-user basis
;
accountcode=lss0101
;
; You can fine tune codecs here using "allow" and
"disallow" clauses
; with specific codecs.  Use "all" to represent all
formats.
;
;disallow=all
;allow=all              ; turns on all installed
codecs
;disallow=g723.1        ; Hm...  Proprietary, don't
use it...
allow=g729              ; Always allow GSM, it's cool
:)
allow=alaw              ; Always allow GSM, it's cool
:)
allow=gsm               ; Always allow GSM, it's cool
:)
;
dtmfmode=rfc2833

Also may I ask if anyone has tried sending fax traffic
on the chan_ss7. Can I ask for a guide on how can I
support T.38 in may asterisk box. I am using asterisk
1.2.7.1

Regards,

Leonimar


--- Goke Aruna <goksie at gmail.com> wrote:

> which h323 do you finally use and I will be glag if
> ur config could be
> shared with me.
> 
> i mean *h323.conf
> 
> goksie
> 
> 
> On 4/20/06, Anton <anton.vazir at gmail.com> wrote:
> >
> > In my hundrends of tests of h323 channels, I've
> found oh323
> > (actually less than others) quite unreliable, it
> crashes in
> > several thousands call attempts. And actually I
> dropped
> > using it because the call setup behaviour - while
> seeting
> > up tyje call - I had very long time betwenn the
> moment when
> > caller picks up and callee hears the answer. Up to
> 5
> > seconds and somtimes more passes from sthe session
> start
> > time and audio apperance! Any similar behaviour
> with you?
> > If no I would suggest you to call someone behind a
> satellite
> > ling - than you would definitelly reproduce that.
> >
> > On 20 April 2006 04:54, leonimar cape wrote:
> > > I am using asterisk-oh323 for the h323
> protocol...
> > >
> > > Regards,
> > >
> > > Leonimar
> > >
> > > --- Anton <anton.vazir at gmail.com> wrote:
> > > > A104D may help in that scenario. Or a kind of
> > > > integration of
> > > > the parts of zapata module to an ss7 module to
> do
> > > > echo
> > > > cancelling in asterisk.
> > > >
> > > > Btw: what protocol do you use to connect
> > > > nextone-h323 with
> > > > Asterisk? (so between the Astersik and
> nextone)
> > > >
> > > > On 19 April 2006 08:03, leonimar cape wrote:
> > > > > Hi
> > > > >
> > > > > Can someone give a suggestion on what I
> should do
> > > >
> > > > to
> > > >
> > > > > omit the echo. Here is may scenario
> > > > >
> > > > >      ---  h323   ---   chan-ss7    ---
> > > > >  A---|  |--------|  |--------------|  |----B
> > > > >      ---         ---               ---
> > > > >     Nextone    Asterisk         DMS
> > > > >
> > > > > The calling party can hear every words
> he/she say
> > > > > after 1  to 2 seconds. But the called party
> can
> > > >
> > > > hear A
> > > >
> > > > > with no echo and the quality is clear. I
> have
> > > >
> > > > already
> > > >
> > > > > tried it in both digium (TE410P) and 
> sangoma card
> > > > > (AT104) and the results where the same. Is
> there
> > > >
> > > > any
> > > >
> > > > > way that I can cancel the echo? Any
> particular
> > > > > settings that I have to change in the
> settings of
> > > >
> > > > the
> > > >
> > > > > asterisk?
> > > > >
> > > > > Thanks in advance
> > > > >
> > > > > Regards,
> > > > >
> > > > > Leonimar
> > > > >
> > > > >
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