<div dir="ltr">Just an update; the solution was to create a new CSS for Cal Forward All and Single Number Reach that prefixes 999 on them; we then match a dial peer using different SIP profile that adds the Diversion (the normal SIP profile does not add a diversion); we also run it thru a translation to remove the 999....<div><br></div><div>Just in case for future reference:</div><div><br></div><div><div><br></div><div>voice class sip-profiles 1</div><div> request INVITE sip-header Diversion modify "<sip:(.*)@(.*>)" "<<a href="http://sip:8475551212@10.10.10.10:5060">sip:8475551212@10.10.10.10:5060</a>>" </div><div> request INVITE sip-header Contact modify "<sip:(.*)@(.*>)" "<<a href="http://sip:8475551212@10.10.10.10:5060">sip:8475551212@10.10.10.10:5060</a>>" </div><div>!</div><div>voice class sip-profiles 2</div><div> request INVITE sip-header Diversion modify "<sip:(.*)@(.*>)" "<<a href="http://sip:8475551212@10.10.10.10:5060">sip:8475551212@10.10.10.10:5060</a>>" </div><div> request INVITE sip-header Contact modify "<sip:(.*)@(.*>)" "<<a href="http://sip:8475551212@10.10.10.10:5060">sip:8475551212@10.10.10.10:5060</a>>" </div><div> request INVITE sip-header Diversion add "Diversion: \"8475551212\" <<a href="mailto:sip%3A8475551212@10.10.10.10">sip:8475551212@10.10.10.10</a>>;reason=unconditional;privacy=off;screen=yes" </div><div>!</div></div><div><br></div><div><div><br></div><div>voice translation-rule 12</div><div> rule 1 /^999\(1..........\)/ /\1/</div><div>!</div></div><div><div><br></div><div>voice translation-profile ANIOUT</div><div> translate called 12</div><div>!</div></div><div><div><br></div><div>dial-peer voice 200 voip</div><div> description OUTBOUND Voice SIP calls to VzB</div><div> translation-profile outgoing ANIOUT</div><div> max-conn 200</div><div> destination-pattern 1[2-9]..[2-9]......$</div><div> session protocol sipv2</div><div> session target sip-server</div><div> incoming called-number .</div><div> voice-class sip dtmf-relay force rtp-nte</div><div> voice-class sip early-offer forced</div><div> voice-class sip profiles 1</div><div> no voice-class sip pass-thru content sdp</div><div> dtmf-relay rtp-nte</div><div> codec g711ulaw</div><div> fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco</div><div> no vad</div><div>!</div></div><div><br></div><div><div><br></div><div>!</div><div>dial-peer voice 200000 voip</div><div> description OUTBOUND CFA-SNR Voice SIP calls to VzB</div><div> translation-profile outgoing ANIOUT</div><div> max-conn 200</div><div> destination-pattern 9991[2-9]..[2-9]......$</div><div> session protocol sipv2</div><div> session target sip-server</div><div> incoming called-number .</div><div> voice-class sip dtmf-relay force rtp-nte</div><div> voice-class sip early-offer forced</div><div> voice-class sip profiles 2</div><div> no voice-class sip pass-thru content sdp</div><div> dtmf-relay rtp-nte</div><div> codec g711ulaw</div><div> fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco</div><div> no vad</div><div>!</div></div><div><br></div><div><br></div><div><br></div><div><br></div><div>Jonathan</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Dec 16, 2014 at 3:37 PM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@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="ltr">OK, added the Diversion, and now on some calls, we are getting a 606 Not Acceptable... <div><br></div><div>voice class sip-profiles 1<br></div><div>request INVITE sip-header Diversion add "Diversion: \"<a href="tel:2125551212" value="+12125551212" target="_blank">2125551212</a>\" <<a href="mailto:sip%3A2125551212@1.1.1.1" target="_blank">sip:2125551212@1.1.1.1</a>>;reason=unconditional;privacy=off;screen=yes"<br></div><div><br></div><div><div>IP/2.0 606 Not acceptable</div><div>Via: SIP/2.0/UDP 157.130.97.178:5060;branch=z9hG4bKAF9861B3B</div><div>From: "John Smith" <<a href="mailto:sip%3A7085551212@scn10001.company.globalipcom.com" target="_blank">sip:7085551212@scn10001.company.globalipcom.com</a>>;tag=389FB18-DAE</div><div>To: <<a href="mailto:sip%3A12125551212@scn10001.company.globalipcom.com" target="_blank">sip:12125551212@scn10001.company.globalipcom.com</a>>;tag=1255691601-1418765447466</div><div>Call-ID: <a href="mailto:9B1AA540-84A111E4-B96AFDC8-B7F757A2@157.130.97.178" target="_blank">9B1AA540-84A111E4-B96AFDC8-B7F757A2@157.130.97.178</a></div><div>CSeq: 101 INVITE</div><div>Timestamp: 1418765435</div><div>Content-Length: 0</div></div><div><br></div><div>Any ideas?</div><span class="HOEnZb"><font color="#888888"><div><br></div><div><br></div><div><br></div><div><br></div><div>Jonathan</div><div><br></div><div><br></div></font></span></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Dec 13, 2014 at 4:23 PM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Cool, let me try it.. I didn't know I could add as well (that seems a tad dangerous, but cool)...<span><font color="#888888"><div><br></div><div><br></div><div>Jonathan</div></font></span></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, Dec 13, 2014 at 4:03 PM, NateCCIE <span dir="ltr"><<a href="mailto:nateccie@gmail.com" target="_blank">nateccie@gmail.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div>You just say add instead of modify. </div><div><br></div><div>Now if you add when there is already one then there will be two. </div><div><br></div><div>So I delete then add so there is always the one I want. <br><br>Sent from my iPhone<div><a href="tel:%2B1%20801%20718%202308" value="+18017182308" target="_blank">+1 801 718 2308</a></div></div><div><div><div><br>On Dec 13, 2014, at 2:47 PM, Jonathan Charles <<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div dir="ltr">The problem from Verizon is that there is no diversion header in the message, how do we add one, I am modifying diversion headers but we don't even see a diversion header on these calls... Are they addable?<div><br></div><div><br></div><div>Jonathan</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Dec 12, 2014 at 1:42 AM, Amit Kumar <span dir="ltr"><<a href="mailto:amit3.kum@gmail.com" target="_blank">amit3.kum@gmail.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Hi,</p>
<p dir="ltr">Do we see this message inbound on cube??, then it would be worth asking vz if they need something specific on redirected calls.</p>
<div class="gmail_quote"><div><div>On 12-Dec-2014 10:33 am, "Jonathan Charles" <<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>> wrote:<br type="attribution"></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">CCM - H323 - CUBE - SIP TRUNK - Verizon SIP<div><br></div><div>We are seeing the following SIP errors on CFA/SNR calls:</div><div><br></div><div><p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">SIP/2.0
503 Request Timeout At Proxy</span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">Via:
SIP/2.0/UDP 157.130.97.178:5060;branch=z9hG4bKACD87F86</span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">From:
"PLACE"
<sip:ORIGINAL OUTSIDE CALLING <a href="mailto:PARTY@scn10001.company.globalipcom.com" target="_blank">PARTY@scn10001.company.globalipcom.com</a>>;tag=EB500DAC-20BF</span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">To:
<sip:SNR DESTINATION @<a href="http://scn10001.company.globalipcom.com" target="_blank">scn10001.company.globalipcom.com</a>>;tag=aprqngfrt-9h004400000a6</span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">Call-ID:
<a href="mailto:41B1BE63-80EF11E4-B597FDC8-B7F757A2@157.130.97.178" target="_blank">41B1BE63-80EF11E4-B597FDC8-B7F757A2@157.130.97.178</a></span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">CSeq:
101 INVITE</span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">Timestamp:
1418358981</span></p>
<p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)">Content-Length:
0</span><span style="color:rgb(31,73,125)"></span></p><p class="MsoNormal"><span style="font-family:'Courier New';color:rgb(31,73,125)"><br></span></p><p class="MsoNormal"><br></p><p class="MsoNormal">We have multiple office codes on our SIP trunk, all of them work except this one; errors are consistent.</p><p class="MsoNormal"><br></p><p class="MsoNormal">There is no DIVERSION field in any of the Invites.</p><p class="MsoNormal"><br></p><p class="MsoNormal">Any ideas?</p><p class="MsoNormal"><br></p><p class="MsoNormal"><br></p><p class="MsoNormal"><br></p><p class="MsoNormal">Jonathan</p><p class="MsoNormal"><br></p></div></div>
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