I've setup a trial Asterisk install based on the RH8 install guide
mentioned on this list. (Thanks, Andy!) I've configured two other
working systems with SJPhone software for SIP. However, while I can
call the phones, the communication is choppy. About every three or four
seconds it cuts out briefly and then returns to normal. Perhaps someone
can give me some pointers. My setups are described below:
1. Asterisk server
2.6 Athlon AMD / 512 MB Ram
Using separate network card.
Using built in sound card. (This should only cause problems for voice
mail and prompts, correct?)
2. Client machine 1
1Ghz Celeron, 256mb Ram, etc.
3. Client machine 2
500 Mhz AMD, 96 mb Ram, etc.
They are connected via a BayNetworks 350F-HD switch. (Fully
capable, however, it does not have advanced packet routing/
prioritizing.) Is it practical to assume that the problem is in the
switch? (ie. needing QoS for VOIP packets) Or perhaps I should be
looking somewhere else to resolve this.
I would appreciate any pointers or suggestions offered.
Steve Lorimer wrote:> I've setup a trial Asterisk install based on the RH8 install guide >mentioned on this list. (Thanks, Andy!) I've configured two other >working systems with SJPhone software for SIP. However, while I can >call the phones, the communication is choppy. About every three or four >seconds it cuts out briefly and then returns to normal. Perhaps someone >can give me some pointers. My setups are described below: > >1. Asterisk server >2.6 Athlon AMD / 512 MB Ram >Using separate network card. >Using built in sound card. (This should only cause problems for voice >mail and prompts, correct?) > >2. Client machine 1 >1Ghz Celeron, 256mb Ram, etc. > >3. Client machine 2 >500 Mhz AMD, 96 mb Ram, etc. > > They are connected via a BayNetworks 350F-HD switch. (Fully >capable, however, it does not have advanced packet routing/ >prioritizing.) Is it practical to assume that the problem is in the >switch? (ie. needing QoS for VOIP packets) Or perhaps I should be >looking somewhere else to resolve this. > I would appreciate any pointers or suggestions offered. > > > > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > > >Are the phones able to reinvite ( canreinvite=yes ) ? This for phone to phone conversation. Any Zaptel HW or ZTDummy loaded ? This for interact with * .