Hi Guys,
I got my Asterisk setup and it seems to be working well.
I don’t have any hardware yet so I’m using 2 X SJphone (Softphone). From inside my Network, I can call Voice Mail, Meetme, IVR and each other.
When I use the same settings from a remote location, I can’t seem to connect to Asterisk. I see nothing in the Logs on Asterisk.
I setup my Router to forward:
UDP 5060 - 5065 192.168. 2.200
(Since UDP didn’t work by itself, I also tried adding TCP)
This setup didn’t work so I tried setting the DMZ (Point All ports) to my Linux Box. That didn’t work either.
What should I try next ?
Thanks,
Dave
google sip_nat.conf
make sure your IP at the router is a public and not a non-routed one
setup a stun server in the softphone set the log so it shows and then see what the deal is
on the server you would do a sip debug or just watch the port
ngrep -d any -Wbyline -t port 5060
g2010
October 3, 2008, 12:21pm
3
Have you configured your sip.conf to be “nat aware”? You probably need something like:
[general]fromdomain=blah.com
Thank you very much for the replies guys.
I added the following file:
sip_nat.conf
externip=205.206.207.208
This is the weird part, I was sure that I added #include sip_nat.conf in my sip.conf file (last night) but now the include is missing.
Now, (from inside) I can actually dial sip:500@myexternalip.domain.com and I get an answer. I’ll try today from the Outside and let you know.
Thanks again,
Dave
Hello Again,
I had the following settings in my sip.conf file (not sip_nat.conf)
externip=205.206.207.208
Now my “remote” or “offlan” softfone car register and make calls to all my extensions. (I still have all my router forwards as stated in my previous post.
But when I added: externip=205.206.207.208, the DTMF’s stopped working on my “Local”. When I call the meetme conference application, it just ignores all my DTMF’s.
I tried to change the DTMF options in SIP.CONF :
dtmfmode=rfc2833 or dtmfmode=inband and I get the same results, ignored DTMF’s.
Any ideas?
Thanks,
Dave