[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
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031204/2ca5192a/attachment.htm