[Asterisk-Users] Asterisk behind NAT << How to do it.

Arnold Ligtvoet asterisk at ligtvoet.org
Wed Dec 3 16:58:28 MST 2003


Leif wrote:
> Awesome!  Have you tried the newer patch / diff for 1.259 (which as of
> right now is the newest chan_sip file).  If you goto bugs.digium.com and
> login anonymously and jump to bug 104, then you can get the newest
> patch.  Same instructions as before.

Installed the new patch, no errors here. Ran make and copied chan_sip.o. All
went fine.

> I just updated it to test the new sip.conf structure which is
>
> externip=
> localnet=
> localmask=

Updated my sip.conf to match these settings. The result seems to be better,
yesterday I noticed a slight delay in the setup of the audio channel, the
speaking clock would only start at the second word, this is now gone.

> Still working great for me here!
>
> BTW!   Can you login to the bug tracker and post a comment ?  Thanks!

I do have one strange issue. I have a test setup here which is very simple.
* server and one windows machine. * is connected via ISDN (chan_i4l) to my
home pbx. On my windows machine I installed Diax, SjPhone and SIPPS. The
strange thing I now have is that both skinny clients are able to receive
audio but not send any when I call an extension on my pbx (so external for
*). I first thought it was my mic, but diax is working fine.

I have already been looking at my sip.conf for the extensions but I'm not
sure if this is the problem. Anyway my sip.conf now is :
[general]
disallow=all                    ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw
allow=ilbc
allow=gsm

; for fix 1.259
externip=212.238.144.173
localnet=192.168.0.100
localmask=255.255.255.0

[phone1]
type=friend
host=dynamic
defaultip=192.168.0.2
dtmfmode=inband
mailbox=1000 ; Mailbox for message waiting indicator
context=default
callerid="Me" <2124>
;reinvite=no
;canreinvite=no
;nat=yes
;insecure=yes

I'll wait your reply for the one-way sound 'issue' (probably me!) before
posting to the bugtracker. Hopefully someone has some clue as to why my sip
clients are not able to send sound.

Thanks,
Arnold Ligtvoet.




More information about the asterisk-users mailing list