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

[robert ivanc]

Arnold Ligtvoet wrote: 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. this patch seems to break my GS phones that are connecting to * via NAT. The one before that works ok - 249 or something? They can't connect anymore - get a Not Found error back. Regards,           Robert 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. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users