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

[Goke Aruna]

I have two links in my linkset which make it 60 channels.
what I am asking of is the whether the cause of my asterisk service stopping
is due to bad oh323 or the version of asterisk i am using which is 1.2.5.

I have tested my chan_ss7 using h323callgen and I populated it with over 64
calls though without any voice and its was successfull.

I am using oh323 version 0.7.3, asterisk 1.2.5 and latest version of
chan_ss7.

goksie

On 4/28/06, Tom Chandler <tchandle at eastex.net> wrote:
>
> If you are using SS7 and E1, then one channel is taken for signaling,
> leaving 30 channels for voice.  SS7 will not send more calls than available
> channels.  You should be getting a message on the console stating no CIC was
> available........
>
> Tom Chandler
>
>
> ----- Original Message -----
> *From:* Goke Aruna <goksie at gmail.com>
> *To:* asterisk-ss7 at lists.digium.com
> *Sent:* Friday, April 28, 2006 1:14 PM
> *Subject:* Re: [asterisk-ss7] asterisk oh323 - chan-ss7 echo problem
>
> Can someone suggest any other h323 protocol that could
> use with asterisk instead of oh323?
>
> I had similar problem with oh323 and each time it happens the asterisk
> will stop at a call going above 30 simultaneus calls
>
> However, I read from voip-info and which give oh323 some ratings to oh323.
>
> I used A104D sangoma card with chan_ss7 and no echo.
>
> goksie
>
> On 4/28/06, Anton <anton.vazir at gmail.com> wrote:
> >
> > Dear Jacob,
> >
> > Any chance to give us any timing approximation for a next
> > version?
> >
> > Regards,
> > Anton.
> >
> > On 28 April 2006 17:26, Jacob Tinning wrote:
> > > On Tue, 18 Apr 2006, leonimar cape wrote:
> > > > 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 next version of chan_ss7 will include
> > > enabling/disabling of the zaptel echo-canceller, which
> > > probably will solve your problem.
> > >
> > > > 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?
> > >
> > > The current version of chan_ss7 does not do anything to
> > > avoid echo. It just passes the audio through the
> > > channels.
> > >
> > > The problem is when A calls an old analog phone B. Old
> > > analog phones typically bleed some of the audio back to
> > > the sender.
> > >
> > > Ordinary synchronous telephone-networks doesn't delay the
> > > audio very much ( < 10ms) so the caller will not notice
> > > any echo.
> > >
> > > Unfortunately, ip-networks induce more delay ( > 70 ms )
> > > which will clearly sound as echo.
> > >
> > > To avoid the echo, the only (as far as I know) solution
> > > is to insert some kind of echo-canceller "in" or "to the
> > > right" of Asterisk in your drawing.
> > >
> > > As noted above, the next version of chan_ss7 will start
> > > the zaptel echo-cancellation when a new call is made
> > > (this is configurable, if you don't want
> > > echo-cancellation), and stop it again when the call is
> > > finished.
> > >
> > > Mvh. Jacob
> > _______________________________________________
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> >
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