<div>Hi,</div>
<div> </div>
<div>I use in Production : Asterisk <a href="http://1.2.9.1">1.2.9.1</a></div>
<div> </div>
<div>We Use Asterisk as a SIP Transit Server to record centrally all the calls.</div>
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<div>The call flow would be:</div>
<div>incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone</div>
<div>outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN</div>
<div> </div>
<div>Dial plan in Asterisk is quite simple:</div>
<div>[record]<br>exten => s,1,Set(CALLFILENAME=${TIMESTAMP}-${UNIQUEID})<br>exten => s,n,Set(CALLERID(name)=${CALLERID(name)})<br>exten => s,n,Set(CALLERID(number)=00${CALLERID(number)})<br>exten => s,n,MixMonitor(${CALLFILENAME}.WAV,b)
<br>exten => s,n,Dial(<a href="mailto:SIP/${DNID}@next-hop,30,r">SIP/${DNID}@next-hop,30,r</a>)</div>
<div> </div>
<div>Everything works perfectly, except when the softswitch, or the PSTN sends back RingBack Tone.</div>
<div> </div>
<div>I can see the RTP flow arriving to Asterisk,</div>
<div>but, it seems that Asterisk doesn't forward it to the other party (next-hop).</div>
<div> </div>
<div>Any ideas why ?</div>
<div>How can I bypass this issue ?</div>
<div> </div>
<div>Thanks,</div>
<div> </div>
<div>Jean-Marc</div>