[Asterisk-Users] Asterisk behind NAT << How to do it.
asterisk at ligtvoet.org
Wed Dec 3 16:58:28 MST 2003
> 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
> I just updated it to test the new sip.conf structure which is
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 :
disallow=all ; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
; for fix 1.259
mailbox=1000 ; Mailbox for message waiting indicator
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.
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