<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">I’m experiencing the same problem. I have 2 providers: one works fine the other not. Sipwise setting are the same.<div class=""><br class=""></div><div class="">The non working provider is Digitel; is that the same of yours, Alessandro?</div><div class=""><br class=""></div><div class="">Apparently is a RFC compliance issue</div><div class=""><br class=""></div><div class="">stefano</div><div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On 19 Jan 2016, at 12:34, Marco Teixeira <<a href="mailto:admin@marcoteixeira.com" class="">admin@marcoteixeira.com</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class=""><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">So i think my initial assumption "(...) If you are allowing direct RTP between endpoints (...)" does not verify then.</div><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">My next step would be to take a SIP trace and check what's going on...</div><div class="gmail_extra"><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small"><br class=""></div><div class=""><div class="gmail_signature"><div dir="ltr" class=""><div class=""><font face="tahoma, sans-serif" class=""><br class=""></font></div><div class=""><font face="tahoma, sans-serif" style="background-color:rgb(255,255,255)" class="">---</font></div><div class=""><font face="tahoma, sans-serif" style="background-color:rgb(255,255,255)" class=""><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small;display:inline">B</div>est regards</font></div><div class=""><font face="tahoma, sans-serif" style="background-color:rgb(255,255,255)" class="">Marco<br class=""></font></div><div class=""><span style="background-color:rgb(255,255,255)" class=""><font face="tahoma, sans-serif" class=""><div class="">---</div><br class=""></font></span></div></div></div></div>
<br class=""><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=""><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<div class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class="">That setting was already enabled.<u class=""></u><u class=""></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class=""><u class=""></u> <u class=""></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class="">Alessandro<u class=""></u><u class=""></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class=""><u class=""></u> <u class=""></u></span></p><p class="MsoNormal"><b class=""><span style="font-size:11.0pt;font-family:"Calibri",sans-serif" class="">Da:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif" class=""> 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><div class=""><div class="h5"><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<u class=""></u><u class=""></u></div></div><div class=""><br class="webkit-block-placeholder"></div><div class=""><div class="h5"><p class="MsoNormal"><u class=""></u> <u class=""></u></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif" class="">Goto your peering server for that "Ext number" <u class=""></u><u class=""></u></span></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif" class="">Click Preferences<u class=""></u><u class=""></u></span></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif" class="">and set use_rtpproxy to always<u class=""></u><u class=""></u></span></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif" class="">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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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif;background:white" class="">Best regards</span><u class=""></u><u class=""></u></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif;background:white" class="">Marco</span><u class=""></u><u class=""></u></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif;background:white" class="">---<u class=""></u><u class=""></u></span></p>
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<div class=""><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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<div class=""><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class="">Yes, the “Ext number” gateway refuses RTP from that endpoint for sure.</span><u class=""></u><u class=""></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class=""> </span><u class=""></u><u class=""></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class="">Could Sipwise be set in order to proxy that RTP flow?</span><u class=""></u><u class=""></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d" class=""> </span><u class=""></u><u class=""></u></p><p class="MsoNormal"><b class=""><span style="font-size:11.0pt;font-family:"Calibri",sans-serif" class="">Da:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif" class=""> 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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<div class=""><p class="MsoNormal"> <u class=""></u><u class=""></u></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif" class="">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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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif;background:white" class="">Best regards</span><u class=""></u><u class=""></u></p>
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<div class=""><p class="MsoNormal"><span style="font-family:"Tahoma",sans-serif;background:white" class="">Marco</span><u class=""></u><u class=""></u></p>
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<div class=""><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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<div class=""><p class="MsoNormal">Hi,<u class=""></u><u class=""></u></p><p class="MsoNormal"> <u class=""></u><u class=""></u></p><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><p class="MsoNormal"> <u class=""></u><u class=""></u></p><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><p class="MsoNormal"> <u class=""></u><u class=""></u></p><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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