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

[Tom Chandler]

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 
  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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