[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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