asterisk users - Jul 2004 - HELP: One way audio... continuously and randomly

All,

I seen already threads about one way audio... but never seen anyone 
answered completely on it.

There is a problem, what we are getting, even with stable-1, CVS updates 
in May, June as well as last Saturday (Jul 10, 2004)
[T1/PRI PSTN] <==> [Lucent Legend PBX] <==> [T1/PRI] <==>
[T100P
Asterisk IAX2] <==> [T1 Internet (ISP Verizon => QWest) connected thru 
T100P interfaces (before it was NetOpia T1 router but the same problem 
existed)] <===> [ADSL Internet (ISP: UTEL/Ukraine)] <===> [IAX2: 
Asterisk with TDM400 cards] <===> [Analog phones & SIP phones (Cisco 
79xx & Polycom IP500]


Calling from here and thru [T1/PRI PSTN] to final phones, analog or just 
sip phones, keep dropping calls, but __ALMOST ALWAYS__ called party does 
not hear when calling party hear well.

We tried different settings for IAX - with and without trunking.
I see the traffic goes both ways and counters on the trunks/channels are 
increasing even when no audio in the phone.

Digium G729 codec is in used, the same problem was exiting when tested 
with iLBC & GSM codecs, but sounds like DID NOT exist with G711 codec (ULAW)


PLEASE HELP!!!! At least where should I start look?

Thank you in advice
Vasyl

:( Just getting silence.... Is this mailing list alive at all?


Vasyl Rublyov wrote:
> All,
>
> I seen already threads about one way audio... but never seen anyone 
> answered completely on it.
>
> There is a problem, what we are getting, even with stable-1, CVS 
> updates in May, June as well as last Saturday (Jul 10, 2004)
> [T1/PRI PSTN] <==> [Lucent Legend PBX] <==> [T1/PRI] <==>
[T100P
> Asterisk IAX2] <==> [T1 Internet (ISP Verizon => QWest) connected
thru
> T100P interfaces (before it was NetOpia T1 router but the same problem 
> existed)] <===> [ADSL Internet (ISP: UTEL/Ukraine)] <===>
[IAX2:
> Asterisk with TDM400 cards] <===> [Analog phones & SIP phones
(Cisco
> 79xx & Polycom IP500]
>
>
> Calling from here and thru [T1/PRI PSTN] to final phones, analog or 
> just sip phones, keep dropping calls, but __ALMOST ALWAYS__ called 
> party does not hear when calling party hear well.
>
> We tried different settings for IAX - with and without trunking.
> I see the traffic goes both ways and counters on the trunks/channels 
> are increasing even when no audio in the phone.
>
> Digium G729 codec is in used, the same problem was exiting when tested 
> with iLBC & GSM codecs, but sounds like DID NOT exist with G711 codec 
> (ULAW)
>
>
> PLEASE HELP!!!! At least where should I start look?
>
> Thank you in advice
> Vasyl
>
>
>
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-- 
Thanks and regards,
  Vasyl Rublyov

On Tue, 2004-07-13 at 11:50, Vasyl Rublyov wrote:
> :( Just getting silence.... Is this mailing list alive at all?
> 
Suggestion:

o Move your existing src to an archive folder.

o Download new cvs code and compile.

o Using new default config files, start with default
  codecs and very simple configs.

o Slowly add back your settings and test between each change till
  you find where the problem is. :)

Maybe there is a simple problem somewhere that you will find.
> 
> Vasyl Rublyov wrote:
> 
> > All,
> >
> > I seen already threads about one way audio... but never seen anyone 
> > answered completely on it.
> >
> > There is a problem, what we are getting, even with stable-1, CVS 
> > updates in May, June as well as last Saturday (Jul 10, 2004)
> > [T1/PRI PSTN] <==> [Lucent Legend PBX] <==> [T1/PRI]
<==> [T100P
> > Asterisk IAX2] <==> [T1 Internet (ISP Verizon => QWest)
connected thru
> > T100P interfaces (before it was NetOpia T1 router but the same problem
> > existed)] <===> [ADSL Internet (ISP: UTEL/Ukraine)] <===>
[IAX2:
> > Asterisk with TDM400 cards] <===> [Analog phones & SIP
phones (Cisco
> > 79xx & Polycom IP500]
> >
> >
-- 
respectfully, Joseph - (606) 477-2355 x140
                       ------=============

Joseph,

Already tried this...

And more interesting that this works fine for days/weeks... and randomly 
start giving this problems for hours/days... after that goes away.
Even without restarting asterisk


Joseph wrote:
>On Tue, 2004-07-13 at 11:50, Vasyl Rublyov wrote:
>  
>
>>:( Just getting silence.... Is this mailing list alive at all?
>>
>>    
>>
>Suggestion:
>
>o Move your existing src to an archive folder.
>
>o Download new cvs code and compile.
>
>o Using new default config files, start with default
>  codecs and very simple configs.
>
>o Slowly add back your settings and test between each change till
>  you find where the problem is. :)
>
>Maybe there is a simple problem somewhere that you will find.
>
>  
>
>>Vasyl Rublyov wrote:
>>
>>    
>>
>>>All,
>>>
>>>I seen already threads about one way audio... but never seen anyone 
>>>answered completely on it.
>>>
>>>There is a problem, what we are getting, even with stable-1, CVS 
>>>updates in May, June as well as last Saturday (Jul 10, 2004)
>>>[T1/PRI PSTN] <==> [Lucent Legend PBX] <==> [T1/PRI]
<==> [T100P
>>>Asterisk IAX2] <==> [T1 Internet (ISP Verizon => QWest)
connected thru
>>>T100P interfaces (before it was NetOpia T1 router but the same
problem
>>>existed)] <===> [ADSL Internet (ISP: UTEL/Ukraine)]
<===> [IAX2:
>>>Asterisk with TDM400 cards] <===> [Analog phones & SIP
phones (Cisco
>>>79xx & Polycom IP500]
>>>      
>>>
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