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benedikt,<br>
try putting these (or your version of these) in the sip.conf [general]
heading. it was suggested to me before, that what is general should
go in the general section, what is specific to a particular extension
should go in the specific extension section. also, I put a ton of
options in my sip.conf, things wouldn't work as I expected. I guess I
didn't understand all the options correctly. took it all out, entered
just a few options, whalla, it worked. KISS<br>
<blockquote><tt>[general]</tt><br>
<tt>context=default</tt><br>
<tt>bindport=5060</tt><br>
<tt>bindaddr=0.0.0.0</tt><br>
<tt>srvlookup=yes</tt><br>
<tt>dtmfmode=rfc2833</tt><br>
<tt>videosupport=yes</tt><br>
<tt>maxcallbitrate=384</tt><br>
<tt>nat=yes</tt><br>
<tt>canreinvite=no</tt><br>
<tt>allowsubscribe=yes</tt><br>
<tt>notifyringing = yes</tt><br>
<tt>limitonpeers=yes</tt><br>
<tt>disallow=all</tt><br>
<tt>allow=ulaw</tt><br>
<tt></tt><tt>allow=gsm</tt><br>
</blockquote>
<blockquote><tt>[401]</tt><br>
<tt>type=friend</tt><br>
<tt>callerid=x401 <(800)000-0401></tt><br>
<tt>secret=1234</tt><br>
<tt>qualify=5000</tt><br>
<tt>nat=no</tt><br>
<tt>host=10.10.15.41</tt><br>
<tt>context=inbound-video</tt><br>
<tt>allow=h264</tt><br>
<tt>mailbox=401<br>
<br>
voicemail.conf - don't forget this one!<br>
</tt></blockquote>
<blockquote><tt>401 => 1234,x401
User,<a class="moz-txt-link-abbreviated" href="mailto:root@localhost,6093053234@atext.com">root@localhost,6093053234@atext.com</a></tt><br>
</blockquote>
remember, I reloaded 1.4.0 over *now with libpri, zaptel,
asterisk-addons too.<br>
<br>
my x-lite version is:<br>
<br>
© 2004 Xten Networks, Inc. All rights reserved.<br>
X-Lite release 1103m build stamp 14262<br>
<br>
it has a diagnostic log, you might want to look at that to see what
codecs the endpoints are negociating to send... would be similar to
below. also the * debugging as below.<br>
<blockquote><tt>From: "asterisk"
<a class="moz-txt-link-rfc2396E" href="sip:asterisk@192.168.15.81"><sip:asterisk@192.168.15.81></a>;tag=as6a0be506</tt><br>
<tt>To: <a class="moz-txt-link-rfc2396E" href="sip:301@192.168.15.103"><sip:301@192.168.15.103></a></tt><br>
<tt>Contact: <a class="moz-txt-link-rfc2396E" href="sip:asterisk@192.168.15.81"><sip:asterisk@192.168.15.81></a></tt><br>
<tt>Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:337fa4724c50b8b14743405110cb3e2f@192.168.15.81">337fa4724c50b8b14743405110cb3e2f@192.168.15.81</a></tt><br>
<tt>CSeq: 102 OPTIONS</tt><br>
<tt>User-Agent: Asterisk PBX</tt><br>
<tt>Max-Forwards: 70</tt><br>
<tt>Date: Thu, 15 Mar 2007 01:10:57 GMT</tt><br>
<b><tt>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY</tt></b><br>
<tt>Supported: replaces</tt><br>
<tt>Content-Length: 0</tt><br>
</blockquote>
-and-<br>
<blockquote><tt>--- (15 headers 11 lines) ---</tt><br>
<tt>Sending to 192.168.15.103 : 5060 (no NAT)</tt><br>
<tt>Using INVITE request as basis request -
<a class="moz-txt-link-abbreviated" href="mailto:c88fa1ed-21a5d8f3-ed584162@192.168.15.103">c88fa1ed-21a5d8f3-ed584162@192.168.15.103</a></tt><br>
<tt>Found user '300'</tt><br>
<tt>Found RTP audio format 0</tt><br>
<tt>Found RTP audio format 8</tt><br>
<tt>Found RTP audio format 18</tt><br>
<tt>Found RTP audio format 101</tt><br>
<tt>Peer audio RTP is at port 192.168.15.103:2226</tt><br>
<b><tt>Found description format PCMU for ID 0</tt><br>
<tt>Found description format PCMA for ID 8</tt><br>
<tt>Found description format G729 for ID 18</tt></b><br>
<tt>Found description format telephone-event for ID 101</tt><br>
<b><tt>Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)</tt></b><br>
<tt>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer -
0x1 (telephone-event), combined - 0x1 (telephone-event)</tt><br>
</blockquote>
the above is * debugging peer 301<br>
I would imagine that you would find your h.263(p) codec in your debug
output... this is from a polycom 301, no video codec...<br>
the users.conf file doesn't, IMHO, to be working reliably, your mileage
may vary... that is why I reloaded over with 1.4.0...<br>
hope that helps...<br>
daveC<br>
<br>
<br>
Benedikt Franz wrote:
<blockquote cite="mid20070314071423.300490@gmx.net" type="cite">
<pre wrap="">Hi Dave,
yes, Audio is fine, but no video. And as far as I can see, X-Lite (running 3.0 build 34025 here, all Clients have exactly the same version) supports only the h.263 and h.263p video codecs. But I am not quite sure if I enabled these codecs properly. For *now, I have put the allow-lines into the users.conf, for instance, heres my setup (I cencored out email and secret):
[6510]
fullname = Benedikt Franz
secret = ...
email = ...
cid_number = 6510
zapchan =
context = numberplan-custom-1
hasvoicemail = yes
hasdirectory = yes
hassip = yes
hasiax = yes
hasmanager = yes
callwaiting = yes
threewaycalling = yes
mailbox = 6510
hasagent = no
group =
host = dynamic
registersip = yes
registeriax = yes
allow = h263
allow = h263p
canreinvite = yes
I am not sure if that is a NAT problem, since all users are either on the local area network, or connected through VPN (I have not tested video with those yet, though), however, I will try that.
</pre>
</blockquote>
<br>
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