[Asterisk-Users] Where is the difference sip.conf - Real-time ?

[Ronald Wiplinger]

I have two phones (111 and 112) on a LAN, and I have on a users site a 
phone 333.

phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are identically set-up (just user name = phone number and 
password are different)

sip.conf (for 111 - all remarked lines removed)

[general]
context=default            ; Default context for incoming calls
port=5060            ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes            ; Enable DNS SRV lookups on outbound calls
tos=lowdelay                   ; 
lowdelay,throughput,reliability,mincost,none
maxexpirey=7200        ; Max length of incoming registration we allow
defaultexpirey=3600        ; Default length of incoming/outoing registration
videosupport=yes        ; Turn on support for SIP video
disallow=all            ; First disallow all codecs
allow=ulaw            ; Allow codecs in order of preference
allow=alaw
allow=g729
allow=gsm
rtcachefriends=yes
rtnoupdate=yes
 rtautoclear=yes
externip = 59.14.2.1
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks


[111]
type=friend
username=hotline
secret=I-know-it
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=111 at default
nat=yes
callgroup=1
pickupgroup=1
callerid="Ronald Hotline",<111>
qualify=1000


Real-time for 112:
name=112
callerid="Ronald Hotline",<112>
canreinvite=yes
context=default
dtmfmode=rfc2833
host=dynamic
language=en
mailbox=112 at default
nat=yes
pickupgroup=1
port=5060
qualify=1000
secret=I-know-it
type=friend
username=112
disallow=all
allow=ulaw;alaw;g729;gsm
cancallforward=yes


Which of the settings cause the different behaviour?
Which settings should I change (maybe not related to the problem)?

bye

Ronald