[Asterisk-Users] Asterisk seems to have more jitter than a hardphone with SIP

[Leon Botes]

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 at 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