[Asterisk-Users] Asterisk as SIP Server

[Sergio Serrano Revuelto]

Hey Srs.
	I have a little problem with the next scenario:

	Internal Phone(801)<-->Asterisk(public IP) <--INTERNET-->ADSL
Router<-->Budgetone(716)
	
|-->ADSL Router<-->Budgetone(717)


My sip.conf is the next:
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = outgoing               ; Default for incoming calls
srvlookup = yes         ; Enable SRV lookups on outbound calls
tos=lowdelay
domain=AVANZADA7
maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=120              ; Default length of incoming/outoing
disallow=all                    ; Disallow all codecs
allow=g729
allow=alaw

[716]
type=friend
username=716
fromuser=716
fromdomain=AVANZADA7
secret=716
host=dynamic
allow=g729
allow=alaw
qualify=yes
canreinvite=yes
context=outgoing
dtmfmode=info  
nat=yes

[717]
type=friend
username=717
fromuser=717
fromdomain=AVANZADA7
secret=717
host=dynamic
allow=g729
allow=alaw
qualify=yes
canreinvite=yes
context=outgoing
dtmfmode=info
nat=yes

[801]
type=friend
username=801
fromuser=801
secret=801
fromdomain=AVANZADA7
host=dynamic
defaultip=192.168.0.185
allow=g729
allow=alaw
allow=ulaw
mailbox=801
context=outgoing
canreinvite=no
dtmfmode=info
nat=no

If I make a call from 716 or 717 to 801 have no problem, but If I try
make a call from 717 to 716 or 716 to 717, signalling is correct but
when a side pickup call, these call hangup. In console appear( warning:
detected firewall/NAT type is UDP blocked)


Any idea,


srsergio