[Asterisk-Users] jittered voice over hisax passive card

[Marko Rakar]

this is now getting interesting;

when I do echo test from my mediatrix unit to asterisk it works
correctly
when I do echo test from my isdn4linux adapter it also works correctly
when I connect two mediatrix units through asterisk they work correctly
when I connect my isdn4linux adapter to public ISDN network it also runs
fine

but when I try to call from isdn4linux passive adapter to mediatrix then
voice comming from mdiatrix is clear while my voice from isdn adapter to
mediatrix is broken, cut off or completely garbgled

I am completely baffled by this

----
Sometimes you're the bug, sometimes you're the windshield.

mailto:marko at printel.hr
http://printel.hr  

-----Original Message-----
From: Marko Rakar 
Sent: Tuesday, March 23, 2004 9:51 AM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] jittered voice over hisax passive card


well

I have found out this setting on my mediatrix unit but it did not solve
my problem (I have solved many RTP warnings in asterisk command prompt
though), since I do not have any other codecs to play with I have
ordered single g729 licence and will play with that (if that solves my
problem, although I think this is somehow hardware related problem)

any other suggestions?


----
Sometimes you're the bug, sometimes you're the windshield.

mailto:marko at printel.hr
http://printel.hr  

-----Original Message-----
From: Rich Adamson [mailto:radamson at routers.com] 
Sent: Monday, March 22, 2004 5:40 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] jittered voice over hisax passive card


> I use mediatrix gateways running SIP protocol.
> 
> I have installed hisax compatible passive adapter on my asterisk box
> (HFC-S PCI Active chip).
> 
> Problem is following; when I dial through my ISDN adapter and run echo

> test I got excellent response (clear sound, no breaks), when I connect

> my SIP gateways between each other users hear each other perfectly, no

> jitters, errors or breaks.
> 
> But; when I try to call from ISDN to SIP gateway I can hear perfectly
> what is said to me from SIP side, but my voice "recorded" on isdn 
> adapters appears jittered or broken to the other party, and if I speak

> to loud it is cut completely.

There is an option in the Mediatrix called Voice Detection (or something
like that) that is set to Auto. Turn that "off".

The problem relates to asterisk needs a constant flow of rtp traffic
(not just traffic when you are speaking), and with the voice detection
feature turned on, asterisk does not get that constant flow.

Rich


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