Leon Botes
2004-Sep-30 10:21 UTC
[Asterisk-Users] Asterisk seems to have more jitter than a hardphone with SIP
I have an asterisk Redhat 9 box running 4 hardphone extensions. Inter-extension calls are crystal clear. However when I dial out through my iconnect account I get a lot of jitter. At first I thought it was my nat gateway but after I programmed on of the hardphones (budge tone 100) for direct dial to iconnect I have clear voice transmission. I have no way of explaining this. My asterisk sip.conf [general] port = 5060 bindaddr = 192.168.255.33 disallow=all allow=ulaw allow=alaw context=bogon-calls [iconnect] context=from-sip type=peer secret=******* username=35205*** host=natrelay.deltathree.com dtmf=rfc2833 callerid="Me" <35205***> canreinvite=no nat=yes [2000] type=friend username=2000 secret=***** host=dynamic defaultip=192.168.255.54 context=from-sip mailbox=2000@local dtmfmode=info callerid="Me" <2000> My extensions.conf [general] static=yes writeprotect=yes [bogon-calls] exten => _.,1,Congestion [from-sip] exten => 2000,1,Dial(SIP/2000,30) exten => 2000,2,Voicemail(u2000) exten => 2000,102,Voicemail(b2000) exten => 2000,103,Hangup exten => _9.,1,Dial(SIP/${EXTEN:1}@iconnect,60,r) exten => 2999,1,VoicemailMain(${CALLERIDNUM}) Can anyone tell me if there is some error I am overlooking. I can't see it being bandwidth or the nat gateway since the budgetone is on the same lan and uses the same gateway. My codec priorities on the hardphone are ulaw then alaw. Thanks in advance. Leon