<div dir="ltr">Yeah, did that too.. and still seeing the unmodified P-Asserted Identity...</div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Mar 16, 2016 at 11:27 AM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@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">If you apply it under voice service voip > sip, then it's applied to all outgoing messages, regardless of dial-peer. Specifying it on the dial-peer level is redundant at that point, though it does take precedence over the global configuration.<div><br></div><div>You can debug ccsip mess and grab an outgoing INVITE example and paste it into the tool I linked before.</div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Mar 16, 2016 at 9:32 AM, 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">Thanks, I checked and the voice class sip profile is on every dial peer not pointing at CCM, it is also applied under voice service voip - sip... <div><br></div><div>I will run the debug after hours to see why it is not being applied...</div><span><font color="#888888"><div><br></div><div><br></div><div><br></div><div>Jonathan</div></font></span></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Mar 16, 2016 at 12:47 AM, Anthony Holloway <span dir="ltr"><<a href="mailto:avholloway+cisco-voip@gmail.com" target="_blank">avholloway+cisco-voip@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"><div>Hey Jonathan,</div><div><br></div>Make sure that the profile is on the correct outgoing dial-peer. SIP profiles are only executed in the outgoing dial-peer, after the dial-peer has been matched/selected.<div><br></div><div>You can use debug voip ccapi inout or debug voip dialpeer to find out which outgoing dial-peer is being matched. Also, you can use debug ccsip all to see if and what the SIP profile is doing.<br><div><br></div><div><div>Assuming dial-peer 200 is your guys, then you can use this tool to help you design your profile:<br></div><div><br></div><div><a href="http://www.cisco.com/web/tsweb/tools/sip-profile/index.html" target="_blank">http://www.cisco.com/web/tsweb/tools/sip-profile/index.html</a><div><br></div><div>I pasted your profile into the tool and gave it a sample input (it's brief, I know) and here is the result:</div><div><br></div><div><img src="cid:ii_1537de4fb55e811a" alt="Inline image 1" style="margin-right:0px"><br></div><div><br></div><div>You should note that the red indicates a match and a replace happened, but also note that the replaced pattern contains an extra > at the end. The reason is because you told it the match ended with .com and not .com>. The reason the first < was replace appropriately, is because your (.*) accounts for it, as well as the space before it.</div><div><br></div><div>I made a small adjustment to your profile, mainly to shorten it up, but to also cleanup the extra > at the end</div><div><br></div><div><img src="cid:ii_1537de7e580ef461" alt="Inline image 2" style="margin-right:0px"><br></div></div></div><div><br></div><div>I've never tried this next method before, but it seems a little cleaner (less regex) than the above two modify profiles, albeit it takes two steps to execute:</div><div><br></div><div><img src="cid:ii_1537df1f2666c470" alt="Inline image 3" style="margin-right:0px"><br></div><div><br></div><div>That last option may not actually work in a real IOS device, as I'm not sure if the message is re-read from the top after a rule matches.</div></div><div><br></div><div>I hope that helps.</div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Wed, Mar 16, 2016 at 12:12 AM, Jonathan Charles <span dir="ltr"><<a href="mailto:jonvoip@gmail.com" target="_blank">jonvoip@gmail.com</a>></span> wrote:<br></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">So, we have a Century Link SIP trunk and we need to send a specific P-Asserted Identity for the call to work.<div><br></div><div>We have: a </div><div><br></div><div>voice class sip-profiles 101<br></div><div><div><br></div><div> request INVITE sip-header P-Asserted-Identity modify "P-Asserted-Identity:(.*)@<a href="http://voip.centurylink.com" target="_blank">voip.centurylink.com</a>" "P-Asserted-Identity: <<a href="mailto:sip%3A3125551212@voip.centurylink.com" target="_blank">sip:3125551212@voip.centurylink.com</a>>" </div><div>!</div></div><div><br></div><div>TO change the P-Asserted ID to what CL wants... however it is not making the change...</div><div><br></div><div><div><br></div><div>dial-peer voice 200 voip</div><div> voice-class sip asserted-id pai</div> voice-class sip profiles 101</div><div><br></div><div>What do we need to do to force a specific P-Asserted Identity...?</div><div><br></div><div><br></div><div>TIA!</div><span><font color="#888888"><div><br></div><div><br></div><div><br></div><div>Jonathan</div></font></span></div>
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