No subject

wrong with the opensips configuration ( since it works from xlite and
not from Grandstream). Maybe there is something wrong with the phones
configuration. Have you set correctly the outbound proxy in
Grandstream? I suggest you to monitor the traffic at the server and
check which are the SIP messages received from Grandstream.<br>
<br>
Regards,<br>
<pre class="moz-signature" cols="72">-- 
Anca Vamanu
<a class="moz-txt-link-abbreviated" href="http://www.voice-system.ro">www.voice-system.ro</a></pre>
<br>
<br>
On 09/26/2010 11:30 PM, misme Gazeta.pl wrote:
<blockquote
 cite="mid:AANLkTimg-FUo87TNH_++WG3yCs0AFYRtRNUn8F3mmJK- at mail.gmail.com"
 type="cite"><span class="Apple-style-span"
 style="font-family: arial,helvetica,clean,sans-serif; font-size: 13px; color: rgb(94, 94, 94); line-height: 16px; white-space: pre-wrap;">I
just have installed opensips and I'm tring to configure it to make
calls like in this scenario:<br>
  <br>
sips registered user -&gt; sips -&gt; asterisk -&gt; sips -&gt; sips
registered user<br>
(I need asterisk to make transcode and bill call).<br>
  <br>
I have used nathelper.cfg config from example with some modifications:<br>
a) I have add modparam("nathelper", "rtpproxy_sock",
"/var/run/rtpproxy.sock",<br>
b) also every time when in config is "route(1);" i have change it to:<br>
  <code
 style="border-style: solid; border-color: rgb(221, 221, 221) rgb(221, 221, 221) rgb(221, 221, 221) rgb(255, 204, 170); border-width: 1px 1px 1px 4px; margin: 1em; padding: 1em 1em 1em 0.5em; overflow: auto; font-style: inherit; font-weight: inherit; font-family: monospace; line-height: 15px; background-color: rgb(221, 221, 221); display: block;"><br>
if(src_ip == 'IP_OF_MY_ASTERISK'){<br>
route(1);<br>
else{<br>
route(2);<br>
}<br>
}<br>
  </code><br>
  <br>
route(2) is:<br>
  <code
 style="border-style: solid; border-color: rgb(221, 221, 221) rgb(221, 221, 221) rgb(221, 221, 221) rgb(255, 204, 170); border-width: 1px 1px 1px 4px; margin: 1em; padding: 1em 1em 1em 0.5em; overflow: auto; font-style: inherit; font-weight: inherit; font-family: monospace; line-height: 15px; background-color: rgb(221, 221, 221); display: block;"><br>
force_rtp_proxy();<br>
rewritehostport("IP_OF_MY_ASTERISK:5060");<br>
t_relay();<br>
  </code><br>
  <br>
so I expect that when I make call to sips registered user from other
than asterisk IP, it will be switched to asterisk (and then asterisk
swtich back to sips and then to user) in other case it will connect to
sips registered user, but it not works every time.<br>
  <br>
I have tested in like this:<br>
X-Lite = sips user 1 (my local IP)<br>
Grandstream HT502 gateway = sips user 2 (my local IP - same as X-Lite)<br>
SIPS - on public IP<br>
Asterisk - on public IP (diferent than SIPS, but on the same server)<br>
  <br>
When I make call:<br>
X-Lite -&gt; Grandstream (via sips) it works fine<br>
  <br>
but when I make call:<br>
Grandstream -&gt; X-Lite (via sips) it dosnt goes throu asterisk (in
asterisk logs there is no info about this), also there is one way audio
from granstream to x-lite (and no audio from x-lite to grandstream).<br>
  <br>
Do you have any idea what is the problem?</span>
  <pre wrap="">
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</blockquote>
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