[Asterisk-Users] jittered voice over hisax passive card

Marko Rakar Marko at printel.hr
Mon Mar 22 08:57:52 MST 2004


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


----
I don't believe you can do today's job with yesterday's methods and be
in business tomorrow. 

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



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