Make the card stop sharing it's IRQ with your IDE controller. Try moving the card to another slot.
Asterisk has to send an audio packet every 20ms for VoIP calls. I believe Zaptel expects no more than a few ms of latency. If something is causing a delay, like the IDE controller locking interrupts and doing disk activity then you're not going to get your interrupts serviced fast enough and you will have audio issues. Doug Crompton wrote: > I am not sure who all see's this list but I do have a few questions that > probably only the developers or somone really in the know of Asterisk > could answer. > > - What is the requirement for timing vs. audio playback in Asterisk. > Specifically voicemail and IVR's (Not meetme) > > - Has this requirment changed in newer versions? > > This obviously is when using Asterisk with no internal cards. I used > Asterisk for several years with a P3 Linux system, NO timing, and it > worked flawlessly. Now with this new Pentium Dual core system I do not > have the perfect audio I experienced with the less powerful system. > > I fully know there are MANY variable here. It could be a combination of > many things, including the OS (Linux Kernel) etc. BUT I offer this input, > Music on Hold works fine. This uses mpg123. So why can this palyback fine > and the other wav/gsm audio be choppy? > > I would gladly switch to a newer Asterisk (using 1.2.29) if someone said > this was solved in that version. > > My system can obviously play (mpg123 - background) audio fine. Why then > does Asterisk internal audio not also play well? > > Doug > > **************************** > * Doug Crompton * > * Richboro, PA 18954 * > * 215-431-6307 * > * * > * [EMAIL PROTECTED] * > * http://www.crompton.com * > **************************** > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
