[Asterisk-Users] SER & Asterisk

[asterisk at geek.be]

Thanks guys, thought SER had to 'register' to be able to use
any Asterisk contexts.
But just defining a new entry in the sip.conf with just context & ip worked!

But now i'm stumbling on another problem.. Asterisk seems to want
to send the SIP udp packets directly to the SIP clients.
In the case of a SIP user/client behind a NAT, this obviously doesn't
work. 

SER is configured to use the wonderful RTPProxy + SER nathelper module, 
and this works flawlessly (using the rewritehostport function).

But when I try to call a phone number on the PSTN network from a SIP
client behind NAT, SER sends the invites to Asterisk, and Asterisk
makes an outbound call to the phone number, the phone rings, but when
the pstn user picks up the phone, no sound, and after a while (couple of
seconds), the call is dropped.
Asterisk spews out the following warning,
  chan_sip.c:471 retrans_pkt: Maximum retries exceeded on call 0c2f021a-7b4d-0133-d19b-7b1794308318 at 192.168.0.5 for seqno 29898 (Response)
Tried searching on the voip-info wiki and mailinglists, but didn't find
a way to force Asterisk to use a SIP proxy/SER.

Any ideas ?


On Fri, Jan 16, 2004 at 12:12:14AM -0800, Chris Albertson wrote:
> 
> Yes, you can keep non-authorized SIP callers from accessing the
> PSTN by setting up the .conf file "correctly" as below
> but you can also
> run a fire wall on the box that Asterisk runs on.  Firewall off
> SIP ports except for if they come from your SER server.
> 
> 
> --- Fran Boon <flavour at partyvibe.com> wrote:
> > [ser]
> > context=sip-legal
> > host=y.y.y.y ; IP address of SER
> > 
> > Se this Wiki page for more flesh of my (not yet fully working!)
> > configs:
> > http://voip-info.org/wiki-Asterisk+cisco+FXO