vicidial.org

I'm doing recordings on all outgoing calls from SIPACCOUNT2. I've put the Monitor command so that even manual calls get recorded. Here's my dialplan:

exten => _62761XXXXXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _62761XXXXXXXXXX,n,Set(CALLERID(number)=8004191068)
exten => _62761XXXXXXXXXX,n,Monitor(wav|${TIMESTAMP}_${EXTEN:5}_OUTBOUND_${CALLERIDNAME}|b)
exten => _62761XXXXXXXXXX,n,Dial(${SIPACCOUNT2}/${EXTEN},30,Ttor)
exten => _62761XXXXXXXXXX,n,congestion()
exten => _62761XXXXXXXXXX,n,Hangup()

The thing is, most of my recordings get out of sync after a while. The recorded file plays fine at the beginnning then gets horribly out of sync as it progress. You can't discern the flow of conversation anymore.

I've recompiled sox thinking it has something to do w/ the mixing. I still get the same results.

All the calls made are on manual mode. It's not passing through Vicidial. Load average is very low since we only have 15 agents. It's not even reaching 1.0 average. Codec used is g729.

I'm using :
asterisk 1.2.14
astguiclient 2.0.2
sox 12.17.9
centos 4.4
athlon x2 Dual Core Processor 3800+
2 gig of ram
80 gig sata

Anyone can shed light on this?

zaptel timer?

i'm using ztdummy. in my other asterisk/vicidial boxes, recordings don't go out of sync. those boxes also uses ztdummy as timer. correct me if i'm wrong, the zaptel timer is primarily used for the meetme conferences right? does it also affect recordings?

the timer is what keeps everything in sync, if it goes out of sync, so can your recordings, especially if they take place in a meetme room.

that's what's puzzling me. the recordings are all from manual calls without any meetme conferences.

I don't really record much that isn't in VICIDIAL, have you tried posting on the asterisk-users list or the digium forums?

try using mixmonitor instead of monitor application in manual dialing context and see if audio is going out of sync in that as well ..