<div dir="ltr">Looks good to me. Might want to pull the CallManager traces to see if the call comes in after going off-hook okay. I can look at them if you want to throw them up on dropbox or something. Sounds like it's doing something now at least.</div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 13, 2015 at 5:15 PM, Barry Howser <span dir="ltr"><<a href="mailto:bhowser5050@gmail.com" target="_blank">bhowser5050@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><div><div>Brian,<br><br></div>I have attached the screen shot of the sip dial rule.<br><br></div>I have the ATA187 using the same CSS on the device and line. That CSS only accesses one partition. That partition has one translation pattern, with a "blank" pattern field and the digits 9911 in the "Called Party Transformation" field. The translation pattern uses a CSS that has access to a 9.911 route pattern (pattern discards predot).<br><br></div>thanks<br></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 13, 2015 at 4:58 PM, Brian Meade <span dir="ltr"><<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</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">Sorry, it was the ATA187s I tried this on. Can you attach a screenshot of your dial rule config?</div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 13, 2015 at 4:15 PM, Brian Meade <span dir="ltr"><<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</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">Right, that's correct. Add 2 PLARs to the SIP Dial Rule with descriptions both with just a button parameter.<div><br></div><div>I've used this for ATA 188s but haven't tested specifically on the 190.</div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Mar 13, 2015 at 3:46 PM, Barry Howser <span dir="ltr"><<a href="mailto:bhowser5050@gmail.com" target="_blank">bhowser5050@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><div><div>hi Brian,<br><br></div>So what you're saying is that in the SIP dial rule; I'll click the "Add Plar" button and then give my parameter a description, select "Button" as my dial parameter then in the value box I'd enter a "1" or a "2" depending on if I wanted the <i>PLAR</i> working on line 1 or 2 of the ATA.<br><br></div>I would then assume that if I wanted both ATA lines to plar, I would have two parameters in the SIP dial rule?<br><br></div>Oyyyy ..... I wish you would write Cisco docs .... I can understand you, lol.<br><br></div><div class="gmail_extra"><br><div class="gmail_quote"><span>On Fri, Mar 13, 2015 at 3:38 PM, Brian Meade <span dir="ltr"><<a href="mailto:bmeade90@vt.edu" target="_blank">bmeade90@vt.edu</a>></span> wrote:<br></span><div><div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">For the SIP Dial Rule, all you want it to have is a PLAR with Button 1 set. Don't enter the number you want to PLAR to. Then just set up PLAR like you would for a SCCP phone with a new CSS/partition/blank translation pattern.</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Fri, Mar 13, 2015 at 3:30 PM, Barry Howser <span dir="ltr"><<a href="mailto:bhowser5050@gmail.com" target="_blank">bhowser5050@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"><div><div><div><div><div>Hello everyone.<br><br></div>I have an ATA190 that needs to do a plar to 911. My dial plan uses "9" to access an outside line (including the 911 pattern).<br><br></div>I created a SIP dial rule and added a plar pattern. I added a parameter called "911" in the description and then added 9911 in the value field. I saved, applied config and restarted.<br><br></div>I have applied that SIP Dial Rule to the ATA190 device's sip dial rule section and reset the ATA. When I take either of the lines off hook with an analog phone, I just get dial tone .... no PLARing.<br><br></div>What am I doing wrong?<br><br></div>thanks<br></div>
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