<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><div><div><div><div><div><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:0 0 0 .8ex;border-left:1px #ccc solid;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><div class="h5"><br>
On 5 Aug 2013, at 06:29, Peter <<a href="mailto:eidevm5@gmail.com">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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