<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; ">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><div><br></div><div>Anthony</div><div><br></div><div><br><div><div>On 7 Aug 2013, at 08:12, 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><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">http://pastebin.freeswitch.org/21274</a><br><br></div>The call is coming from <a href="mailto:1001@10.1.1.204">1001@10.1.1.204</a> to <a href="mailto:3000@10.10.10.203">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:0 0 0 .8ex;border-left:1px #ccc solid;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 class="HOEnZb"><font color="#888888"><div><span><br>
</span></div><div><span>Anthony </span></div></font></span><div><div class="h5"><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: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><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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