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<div>No, i'm not using Digitel. Also I don't think that we are talking about the same issue.</div>
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<span style="font-size:11.0pt; font-family:'Calibri','sans-serif'"><b>Da:</b> stefanormc <stefanormc@gmail.com><br>
<b>Inviato:</b> 19/gen/2016 12:36<br>
<b>A:</b> Marco Teixeira<br>
<b>Cc:</b> Alessandro Bolletta; spce-user@lists.sipwise.com<br>
<b>Oggetto:</b> Re: [Spce-user] Call forwarding, no RTP<br>
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<div>I’m experiencing the same problem. I have 2 providers: one works fine the other not. Sipwise setting are the same.
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<div class="">The non working provider is Digitel; is that the same of yours, Alessandro?</div>
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<div class="">Apparently is a RFC compliance issue</div>
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<div class="">stefano</div>
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<div class="">On 19 Jan 2016, at 12:34, Marco Teixeira <<a href="mailto:admin@marcoteixeira.com" class="">admin@marcoteixeira.com</a>> wrote:</div>
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So i think my initial assumption "(...) If you are allowing direct RTP between endpoints (...)" does not verify then.</div>
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My next step would be to take a SIP trace and check what's going on...</div>
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est regards</font></div>
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<div class="gmail_quote">On Tue, Jan 19, 2016 at 11:24 AM, Alessandro Bolletta <span dir="ltr" class="">
<<a href="mailto:alessandro@mediaspot.net" target="_blank" class="">alessandro@mediaspot.net</a>></span> wrote:<br class="">
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<p class="MsoNormal"><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif; color:#1f497d">That setting was already enabled.<u class=""></u><u class=""></u></span></p>
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<p class="MsoNormal"><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif; color:#1f497d">Alessandro<u class=""></u><u class=""></u></span></p>
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<p class="MsoNormal"><b class=""><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif">Da:</span></b><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif"> Marco Teixeira [mailto:<a href="mailto:admin@marcoteixeira.com" target="_blank" class="">admin@marcoteixeira.com</a>]
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<b class="">Inviato:</b> martedì 19 gennaio 2016 12.12</span></p>
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<b class="">A:</b> Alessandro Bolletta <<a href="mailto:alessandro@mediaspot.net" target="_blank" class="">alessandro@mediaspot.net</a>><br class="">
<b class="">Cc:</b> <a href="mailto:spce-user@lists.sipwise.com" target="_blank" class="">
spce-user@lists.sipwise.com</a><br class="">
<b class="">Oggetto:</b> Re: [Spce-user] Call forwarding, no RTP<u class=""></u><u class=""></u></div>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif">Goto your peering server for that "Ext number" <u class=""></u><u class=""></u></span></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif">Click Preferences<u class=""></u><u class=""></u></span></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif">and set use_rtpproxy to always<u class=""></u><u class=""></u></span></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif">I think that should do make the RTP for calls to that peer to always proxy via SPCE<u class=""></u><u class=""></u></span></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif; background:white">Best regards</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif; background:white">Marco</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal">On Tue, Jan 19, 2016 at 10:37 AM, Alessandro Bolletta <<a href="mailto:alessandro@mediaspot.net" target="_blank" class="">alessandro@mediaspot.net</a>> wrote:<u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif; color:#1f497d">Yes, the “Ext number” gateway refuses RTP from that endpoint for sure.</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif; color:#1f497d">Could Sipwise be set in order to proxy that RTP flow?</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><b class=""><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif">Da:</span></b><span class="" style="font-size:11.0pt; font-family:"Calibri",sans-serif"> Marco Teixeira [mailto:<a href="mailto:admin@marcoteixeira.com" target="_blank" class="">admin@marcoteixeira.com</a>]
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<b class="">Inviato:</b> martedì 19 gennaio 2016 10.51<br class="">
<b class="">A:</b> Alessandro Bolletta <<a href="mailto:alessandro@mediaspot.net" target="_blank" class="">alessandro@mediaspot.net</a>><br class="">
<b class="">Cc:</b> <a href="mailto:spce-user@lists.sipwise.com" target="_blank" class="">
spce-user@lists.sipwise.com</a><br class="">
<b class="">Oggetto:</b> Re: [Spce-user] Call forwarding, no RTP</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif">If you are allowing direct RTP between endpoints, your "external number" gateway/provider might be refusing RTP from the original endpoint and expecting audio to come from your SPCE.</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif; background:white">Best regards</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal"><span class="" style="font-family:"Tahoma",sans-serif; background:white">Marco</span><u class=""></u><u class=""></u></p>
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<p class="MsoNormal">On Mon, Jan 18, 2016 at 4:09 PM, Alessandro Bolletta <<a href="mailto:alessandro@mediaspot.net" target="_blank" class="">alessandro@mediaspot.net</a>> wrote:<u class=""></u><u class=""></u></p>
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<p class="MsoNormal">Hi,<u class=""></u><u class=""></u></p>
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<p class="MsoNormal">if I configure a subscriber to unconditionally forward any incoming call to an external number, get the forwarded call estabilished but I get no RTP flows on both sides.<u class=""></u><u class=""></u></p>
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<p class="MsoNormal">The peer used for the call forwarding is already working flawlessly for other purposes, so I can imagine that there’s something with call forwarding’s procedure.<u class=""></u><u class=""></u></p>
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<p class="MsoNormal">Hoping to get your help,<u class=""></u><u class=""></u></p>
<p class="MsoNormal">Alessandro Bolletta<u class=""></u><u class=""></u></p>
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