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Arnold Ligtvoet wrote:<br>
<blockquote type="cite"
cite="midHJEMIJNKGELBKDGINJGMEEGMCIAA.asterisk@ligtvoet.org">
<pre wrap="">Leif wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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.
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
</blockquote>
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.<br>
<br>
Regards,<br>
<br>
Robert<br>
<br>
<blockquote type="cite"
cite="midHJEMIJNKGELBKDGINJGMEEGMCIAA.asterisk@ligtvoet.org">
<pre wrap="">Installed the new patch, no errors here. Ran make and copied chan_sip.o. All
went fine.
</pre>
<blockquote type="cite">
<pre wrap="">I just updated it to test the new sip.conf structure which is
externip=
localnet=
localmask=
</pre>
</blockquote>
<pre wrap=""><!---->
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.
</pre>
<blockquote type="cite">
<pre wrap="">Still working great for me here!
BTW! Can you login to the bug tracker and post a comment ? Thanks!
</pre>
</blockquote>
<pre wrap=""><!---->
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.
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</pre>
</blockquote>
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