search for: snocetti

hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...sers@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 10 > From: "Sebastian Nocetti" <snocetti@fibertel.com.ar> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] STILL NO AUDIO > Date: Mon, 19 Jul 2004 12:51:49 -0300 > Reply-To: asterisk-users@lists.digium.com > > Testing both... > > -----Mensaje original----- > De: asterisk-users-adm...

Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...

I tried a lot of times to get it worked, but I cant obtain audio using SIP<->chan_h323 or chan_h323<->SIP I tried disbling FastStart without good results... What's the problem? I need to do BRIDGE between SIP and H.323!! help!! Sebastian.- -------------- next part -------------- An HTML attachment was scrubbed... URL:

is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working????? I am using chan_h323 Thanks!! Sebastian Nocetti. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: