<html><body><div style="font-family: arial, helvetica, sans-serif; font-size: 12pt; color: #000000"><div>First of all .. rtpengine could not 'transcode' ... as it doesn't have that feature.</div><div><br data-mce-bogus="1"></div><div>I think all you whant it's to route all traffic coming from a peer, route it to another peer, in both directions, from peer A to peer B and viceversa, jus doing some rewriting of the request and doing proxing of the RTP throught rtpengine.</div><div><br data-mce-bogus="1"></div><div>Well, that's easy ... you only need to define the two peers, the rewriting rules you whant ... and set the option inside the peers definition that let you route call from one peer, back to another peer.</div><div><br></div><hr id="zwchr" data-marker="__DIVIDER__"><div data-marker="__HEADERS__"><blockquote style="border-left:2px solid #1010FF;margin-left:5px;padding-left:5px;color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><b>De: </b>"Manu Raghavan" <manu@voxology.co><br><b>Para: </b>spce-user@lists.sipwise.com<br><b>Enviados: </b>Sábado, 21 de Marzo 2015 15:04:54<br><b>Asunto: </b>[Spce-user] SPCE as an SBC between proxy and trunk<br></blockquote></div><div data-marker="__QUOTED_TEXT__"><blockquote style="border-left:2px solid #1010FF;margin-left:5px;padding-left:5px;color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;"><div>
Hello,
</div><br><div>I’m having trouble setting up Sipwise as an SBC between a SIP trunk, and another SIP proxy.</div><div>The goal is to have spce’s rtpengine transcode media between 2 peers, a trunk and a proxy.</div><br><div>I’ve tried multiple configurations, including peerings, domains, and subscribers with permanent registrations, but nothing seems to work quite right. In most cases, the call gets routed inwards, but never onward to a peer. I’m not quite sure if I do at all need to set up (A) a domain, and (B) subscribers, since all I’m really trying to do is set transcoding rules between 2 SIP peers.</div><br><div>Past discussions on your mailing list seem to indicate that I only need 2 peering servers, and rules to match them. The trouble seems to be that no calls never leave the kamailio proxy, with something to the tune of “Callee is not local” followed by "Call to unknown local user from unauthenticated foreign domain.” Really, I don’t need or want any users (i.e. subscribers) at all.</div><br><div>Any guidance would me much appreciated.</div><br><div>Thanks in advance.</div><br><div>Manu</div>
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