<div dir="ltr"><div><div><div><div><div><div>Hi Anthony<br><br></div>Really appreciate the taking your time to look at this.  It&#39;s starting to drive me nuts.<br><br></div>My dialplan for calls to CME is:<br><br><span style="font-family:courier new,monospace">  &lt;extension name=&quot;incoming-cme&quot;&gt;<br>
        &lt;condition field=&quot;destination_number&quot; expression=&quot;^(3\d\d\d)$&quot;&gt;<br>            &lt;action application=&quot;bridge&quot; data=&quot;sofia/internal/${<a href="mailto:destination_number%7D@10.10.10.203">destination_number}@10.10.10.203</a>&quot;/&gt;<br>
        &lt;/condition&gt;<br>  &lt;/extension&gt;</span><br><br><br></div>My internal sip profile has:<br><span style="font-family:courier new,monospace"><br>    &lt;param name=&quot;rtp-ip&quot; value=&quot;10.10.10.206&quot;/&gt;<br>
    &lt;param name=&quot;sip-ip&quot; value=&quot;10.10.10.206&quot;/&gt;<br>    &lt;param name=&quot;ext-rtp-ip&quot; value=&quot;10.1.1.206&quot;/&gt;<br>    &lt;param name=&quot;ext-sip-ip&quot; value=&quot;10.1.1.206&quot;/&gt;</span><br>
<br></div>The output from:<br><br></div><div>sofia status<br><br></div><div>is:<br><span style="font-family:courier new,monospace"><br>internal    profile                 <a href="http://sip:mod_sofia@10.1.1.206:5060">sip:mod_sofia@10.1.1.206:5060</a>    RUNNING (0)<br>
external    profile                 <a href="http://sip:mod_sofia@10.1.1.206:5060">sip:mod_sofia@10.1.1.206:5060</a>    RUNNING (0)</span><br><br></div><div>Should internal show 10.10.10.206??<br><br></div><div>The output from:<br>
</div><div><br></div>sofia status profile internal<br><br></div>shows:<br><br><span style="font-family:courier new,monospace">Name                 internal<br>Domain Name          N/A<br>Auto-NAT             false<br>DBName               sofia_reg_internal<br>
Pres Hosts           10.10.10.206,10.1.1.206<br>Dialplan             XML<br>Context              public<br>Challenge Realm      auto_from<br>RTP-IP               10.10.10.206<br>Ext-RTP-IP           10.1.1.206<br>SIP-IP               10.10.10.206<br>
Ext-SIP-IP           10.1.1.206<br>URL                  <a href="http://sip:mod_sofia@10.1.1.206:5060">sip:mod_sofia@10.1.1.206:5060</a><br>BIND-URL             sip:mod_sofia@10.1.1.206:5060;maddr=10.10.10.206<br>HOLD-MUSIC           N/A<br>
OUTBOUND-PROXY       N/A<br>CODECS IN            iLBC@30i,PCMU,PCMA,GSM<br>CODECS OUT           iLBC@30i,PCMU,PCMA,GSM<br>TEL-EVENT            101<br>DTMF-MODE            rfc2833<br>CNG                  13<br>SESSION-TO           0<br>
MAX-DIALOG           0<br>NOMEDIA              false<br>LATE-NEG             false<br>PROXY-MEDIA          false<br>ZRTP-PASSTHRU        false<br>AGGRESSIVENAT        false<br>STUN-ENABLED         true<br>STUN-AUTO-DISABLE    false</span><br>
<br><br><div><div><div><div><div><div><div><div class="gmail_extra">The SIP trace from the Freeswitch SBC is at:<br><br><a href="http://pastebin.freeswitch.org/21279">http://pastebin.freeswitch.org/21279</a><br><br></div>
<div class="gmail_extra">I&#39;ve been playing around with all sorts of different combinations of SIP/RTP IP settings, but still no closer.<br><br>Hope you have some insight.<br><br></div><div class="gmail_extra">Thanks<br>
<br>Peter<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <span dir="ltr">&lt;<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div style="word-wrap:break-word">Hi Peter,<div><br></div><div>Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side. </div>
<div>Check your bridge statement, is it using the correct sip profile? </div><div>Check your sip profile SBC internal params rtp-ip &amp; sip-ip, make sure they are set correctly to 10.10.10.206</div><div>Paste up your logs from the sbc including sip trace.</div>
<span class=""><font color="#888888"><div><br></div><div>Anthony</div></font></span><div><div class="h5"><div><br></div><div><br><div><div>On 7 Aug 2013, at 08:12, Peter &lt;<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>&gt; wrote:</div>
<br><blockquote type="cite"><div dir="ltr"><div><div><div><div><div><div><div><div><div><div>Hi Anthony.<br><br></div>Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.<br><br></div>Note that CME only has one interface, so binding the source interface doesn&#39;t really make much sense.<br>

<br>Note that I&#39;ve simplified my set up a little and the phone that was registered to CUCM is now registered to CME.  However, the result is still the same, ie:  one way audio to the Cisco phone.<br><br></div>You can see the SIP debug from CME at:<br>

<br><a href="http://pastebin.freeswitch.org/21274" target="_blank">http://pastebin.freeswitch.org/21274</a><br><br></div>The call is coming from <a href="mailto:1001@10.1.1.204" target="_blank">1001@10.1.1.204</a> to <a href="mailto:3000@10.10.10.203" target="_blank">3000@10.10.10.203</a><br>

<br></div>where<br><br></div>10.1.1.204  - Freeswitch where SIP clients register to<br></div>10.1.1.206  - External side of Freeswitch SBC<br></div>10.10.10.206 - Internal side of Freeswitch SBC<br></div>10.10.10.203 - CME<br>

<br></div>Peter<br><br><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <span dir="ltr">&lt;<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="auto"><div>Hi Peter,</div><div><br></div><div>Because the calls are fine when using Kamailio I&#39;m assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?</div>

<div><span>Yes you are correct. Have you added the commands? Add them as a first step.</span></div><div><span>Send on a &#39;debug ccsip messages&#39; </span></div><span><font color="#888888"><div><span><br>
</span></div><div><span>Anthony </span></div></font></span><div><div><div><br></div><div><br></div><div><br>On 6 Aug 2013, at 05:35, Peter &lt;<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>&gt; wrote:<br>

<br></div><blockquote type="cite"><div dir="ltr"><div><div><div><div><div><div>Thanks for replying Anthony.<br><br></div>Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.<br>

<br></div>As far as I can see<br>
<br></div>voice-class sip bind media source-interface ....<br><br></div>is just used to bind the SIP or media stream to the appropriate interface on the CUBE.<br><br></div>My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.<br>


<br></div>Is my understanding of the sip bind media command correct?<br><br>Thanks<br><br>Peter<br><div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <span dir="ltr">&lt;<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">On cube make sure you specify the source address on your dial-peers<br>
voice-class sip bind media|control<br>
to the correct side. I have seen one way audio when not set.<br>
<div><br>
On 5 Aug 2013, at 06:29, Peter &lt;<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>&gt; wrote:<br>
<br>
&gt;<br>
&gt;<br>
&gt; I currently have successful two way calls (signalling and media) in the following setup<br>
&gt;<br>
&gt;<br>
&gt; External Linphone --&gt;  Freeswitch  --&gt; Freeswitch SBC  -&gt; Router  -&gt; Kamailio --&gt; Internal Linphone<br>
&gt;<br>
&gt; However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:<br>
&gt;<br>
&gt; External Linphone --&gt;  Freeswitch  --&gt; Freeswitch SBC  -&gt; Router  -&gt; CME -&gt;  CUCM9 --&gt;  Cisco handset<br>
&gt;<br>
&gt; The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.<br>
&gt;<br>
&gt; Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.<br>
&gt;<br>
&gt; Looking at the RTP debugs on CME I can see the problem is that the &quot;Media Dest Addr&quot; is getting set to the external side of the FS SBC rather than the internal IP address.<br>
&gt;<br>
&gt;<br>
&gt; I tried setting adding:<br>
&gt;<br>
&gt;             &lt;action application=&quot;set&quot; data=&quot;disable_rtp_auto_adjust=&quot;true&quot; /&gt;<br>
&gt;<br>
&gt; to the dialplan on the SBC, but it made no difference.<br>
&gt;<br>
&gt;<br>
&gt; Any suggestions as to what to check next?<br>
&gt;<br>
&gt; Thanks<br>
&gt;<br>
&gt; Peter<br>
&gt;<br>
</div>&gt; _________________________________________________________________________<br>
<br>
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