[Asterisk-Users] jittered voice over hisax passive card

[Marko Rakar]

one more thing

I have just configured so that I enter asterisk through ttyI0 and then
exit back to PSTN (or in my case ISDN) thru ttyI1 (second B channel on
the same adapter) and zero problems, sound is perfect, no jittering,
breaks or any problem whatsoever

so something happens in between asetrisk box and my SIP gateway and I
really do not have a clue what

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

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

-----Original Message-----
From: Marko Rakar 
Sent: Monday, March 22, 2004 4:58 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] jittered voice over hisax passive card



I have latest asterisk running on redhat 9;

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.


I use ulaw/alaw on my SIP gateways.

this is my modem.conf file (this is channel one, I have one more
running); msn=340 driver=i4l type=autodetect incomingmsn=340 device =>
/dev/ttyI0

this is my sip.conf file (sample; I have 7 more identical ports);

[201]
type=friend
username=201
host=dynamic
defaultip=192.168.3.210
dtmfmode=inband

any clues, ideas what to check?



p.s. also, when SIP user calls my ttyI0 then I do not hear ringing tone