<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Yes internal should show 10.10.10.206<div>change ext-sip-ip and ext-rtp-ip to "auto-nat" </div><div>Restart and check sofia status</div><div><br></div><div>From the wiki</div><div><div><div>ext-rtp-ip</div><div>This is the IP behind which FreeSWITCH is seen from the Internet, so if FreeSWITCH is behind NAT, this is basically the public IP that should be used for RTP.</div></div><div><br></div><div><div>On 8 Aug 2013, at 02:30, Peter <<a href="mailto:eidevm5@gmail.com">eidevm5@gmail.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><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'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"> <extension name="incoming-cme"><br>
<condition field="destination_number" expression="^(3\d\d\d)$"><br> <action application="bridge" data="sofia/internal/${<a href="mailto:destination_number%7D@10.10.10.203">destination_number}@10.10.10.203</a>"/><br>
</condition><br> </extension></span><br><br><br></div>My internal sip profile has:<br><span style="font-family:courier new,monospace"><br> <param name="rtp-ip" value="10.10.10.206"/><br>
<param name="sip-ip" value="10.10.10.206"/><br> <param name="ext-rtp-ip" value="10.1.1.206"/><br> <param name="ext-sip-ip" value="10.1.1.206"/></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 <a href="sip:mod_sofia@10.1.1.206:5060;maddr=10.10.10.206">sip:mod_sofia@10.1.1.206:5060;maddr=10.10.10.206</a><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 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'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"><<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>></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 & 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 <<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>> 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't really make much sense.<br>
<br>Note that I'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"><<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>></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'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 'debug ccsip messages' </span></div><span><font color="#888888"><div><span><br>
</span></div><div><span>Anthony </span></div></font></span><div><div><br></div><div><br></div><div><br>On 6 Aug 2013, at 05:35, Peter <<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>> 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"><<a href="mailto:agtmcgarry@gmail.com" target="_blank">agtmcgarry@gmail.com</a>></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 <<a href="mailto:eidevm5@gmail.com" target="_blank">eidevm5@gmail.com</a>> wrote:<br>
<br>
><br>
><br>
> I currently have successful two way calls (signalling and media) in the following setup<br>
><br>
><br>
> External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> Kamailio --> Internal Linphone<br>
><br>
> However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:<br>
><br>
> External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> CME -> CUCM9 --> Cisco handset<br>
><br>
> 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>
><br>
> Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.<br>
><br>
> Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.<br>
><br>
><br>
> I tried setting adding:<br>
><br>
> <action application="set" data="disable_rtp_auto_adjust="true" /><br>
><br>
> to the dialplan on the SBC, but it made no difference.<br>
><br>
><br>
> Any suggestions as to what to check next?<br>
><br>
> Thanks<br>
><br>
> Peter<br>
><br>
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