[Asterisk-Users] NAT/Routing problem

[Rudolf Ladyzhenskii]

Hi, all

I have Asterisk here and SIP phone sitting at another location.

Initially, I had problems registering the phone. Now I have added 'nat=yes' 
for this phone in sip.conf and phone registers.
However, I can not make calls.

SIP debug shows that phone registers with public IP address of the site, 
while calls somehow go to local address.

Here is an example of SIP debug message:

-- Registered SIP 'ext102' at 147.10.78.157 port 8103 expires 3600


    -- Attempting native bridge of SIP/ext102-26a4 and SIP/ext101-1b49
Feb 27 17:02:31 WARNING[3160]: chan_sip.c:755 retrans_pkt: Maximum retries 
excee
ded on call b27335a9-ad36406b-b2b8ed30 at 192.168.1.2 for seqno 2 (Non-critical 
Res
ponse)

As one can see, public IP 147.10.78.157 is used at registration time, while 
private IP 192.168.1.2 is used for communicating with phone.

Remote site does not have firewall. My site does, but I could not see 
anything wrong there. I have turned on logging on firewall and no suspicios 
activity goes on.

Any help is appreciated.

Thanks,
Rudolf
P.S. Here is extract from my sip.conf file:

[ext102]
type=user
nat=yes
host=dynamic
secret=ext102
context=default

[ext102]
type=peer
nat=yes
secret=ext102
host=dynamic
context=default
callerid="Ext 102"