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:[EMAIL PROTECTED] http://printel.hr -----Original Message----- From: Marko Rakar Sent: Monday, March 22, 2004 4:58 PM To: [EMAIL PROTECTED] 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
