Hi Andreas, got it working both ways now. Many thanks for nice work.<div><br></div><div>I will now test it further with transcoding.</div><div><br>And start on the documentation.</div><div><br></div><div>Rgds</div><div>Nik</div>
<div><br><div class="gmail_quote">On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar <span dir="ltr">&lt;<a href="mailto:nikpakar@gmail.com">nikpakar@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi Andreas, i now have calls coming from external sip-&gt;LCR-&gt;gsm<div><br></div><div>But still cant figure out the dial plan to send a call out on sip, gsm-&gt;LCR-&gt;sip</div><div><br></div><div>I tried,</div><div><br>

</div><div>dialing=072333444<span style="white-space:pre-wrap">			</span>: extern interfaces=sip prefix=072333444</div><div><br></div><div>But on LCR trace it shows as below. I think im missing some thing small here. Appreciate if you can give a little hint.</div>

<div><br></div><div>Thanks<br>nik</div><div><br></div><div><div>06.03.12 12:40:41.286 EP(1): ACTION (match)  action goto  line 11</div><div>06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to)  ruleset extern  dialing 072333444</div>

<div>06.03.12 12:40:41.286 EP(1): ACTION (match)  action extern  line 28</div><div>06.03.12 12:40:41.286 EP(1): ACTION extern (calling)  number 072333444  interfaces sip</div><div>06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE  to CH(1)</div>

<div>06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface)  interface sip</div><div>06.03.12 12:40:41.287 EP(2): INTERFACE (has no function)  interface�@ </div><div>06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found)</div>

<div>06.03.12 12:40:41.287 EP(1): TONE  to CH(1)  directory default  name cause_22</div><div>06.03.12 12:40:41.287 EP(1): DISCONNECT  to CH(1)  cause value=34 location=1-Local-PBX</div><div>06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR&lt;-&gt;BSC  progress coding=3 location=1 descr=8  cause coding=3 location=1 value=34</div>

<div>06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR&lt;-&gt;BSC  cause coding=3 location=0 value=16</div><div>06.03.12 12:40:56.247 EP(1): RELEASE  from CH(1)  cause value=16 location=0-User</div><div>06.03.12 12:40:56.247 EP(1): ACTION hangup</div>
<div><div class="h5">
<div><br></div><br><div class="gmail_quote">On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar <span dir="ltr">&lt;<a href="mailto:nikpakar@gmail.com" target="_blank">nikpakar@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

Hi Andreas,<div><br></div><div>It was apperently compiled on my debian while i had misdn libs isntalled. Now im trying on a fresh debian from the same set of sources which i got it working and without misdn, it fails to compile gsm.</div>


<div><br></div><div>Attached is my config output and compile break.</div><div><br></div><div><a href="http://pastebin.com/W6UHn4Lc" target="_blank">http://pastebin.com/W6UHn4Lc</a></div><div><br></div><div>So should i still install misdn even though its not used.</div>


<div><br></div><div>Rgds</div><div><span><font color="#888888">Nik</font></span><div><div><br><br><div class="gmail_quote">On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg <span dir="ltr">&lt;<a href="mailto:andreas@eversberg.eu" target="_blank">andreas@eversberg.eu</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>Alexander Chemeris wrote:<br>
&gt; Does it mean that that now we can use LCR with other SIP<br>
&gt; softswitches/PBX&#39;es, like Freeswitch? I do not follow LCR development<br>
&gt; closely, but that would be a very interesting development.<br>
&gt;<br>
</div>yes, this was my intention. gsm and sip interface of lcr will not rely<br>
on misdn anymore. currently i don&#39;t support audio transfer via chan_lcr,<br>
so chan_lcr will only work with isdn interfaces. the sip interface<br>
implementation has not much options, so it can only do sipmple<br>
point-to-point sip connections to a gateway or endpoint.<br>
</blockquote></div><br></div></div></div>
</blockquote></div><br></div></div></div>
</blockquote></div><br></div>