[Asterisk-Users] HELP: One way audio... continuously and randomly

Vasyl Rublyov vasyl.rublyov at ionidea.com
Mon Jul 12 14:53:22 MST 2004


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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