[SOLVED] Here is what ended up working for me;
ATA190 firmware 1.1.2 (posted to CCO on 02-15-2015) - Specifically states PLAR isn't supported until Version 1.1.2 SIP Dial Rule with 1 blank PLAR pattern (no button value) Translation pattern using the *!* wildcard, not "" (blank) Traces showed that when that ATA went off-hook, it wasn't dialing anything. As soon as I used the *!* wildcard instead of "" (blank) in the translation, the ATA would dial it when going off hook. On Fri, Mar 13, 2015 at 5:25 PM, Brian Meade <bmead...@vt.edu> wrote: > 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. > > On Fri, Mar 13, 2015 at 5:15 PM, Barry Howser <bhowser5...@gmail.com> > wrote: > >> Brian, >> >> I have attached the screen shot of the sip dial rule. >> >> 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). >> >> thanks >> >> On Fri, Mar 13, 2015 at 4:58 PM, Brian Meade <bmead...@vt.edu> wrote: >> >>> Sorry, it was the ATA187s I tried this on. Can you attach a screenshot >>> of your dial rule config? >>> >>> On Fri, Mar 13, 2015 at 4:15 PM, Brian Meade <bmead...@vt.edu> wrote: >>> >>>> Right, that's correct. Add 2 PLARs to the SIP Dial Rule with >>>> descriptions both with just a button parameter. >>>> >>>> I've used this for ATA 188s but haven't tested specifically on the 190. >>>> >>>> On Fri, Mar 13, 2015 at 3:46 PM, Barry Howser <bhowser5...@gmail.com> >>>> wrote: >>>> >>>>> hi Brian, >>>>> >>>>> 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 *PLAR* working on line 1 or 2 of the ATA. >>>>> >>>>> I would then assume that if I wanted both ATA lines to plar, I would >>>>> have two parameters in the SIP dial rule? >>>>> >>>>> Oyyyy ..... I wish you would write Cisco docs .... I can understand >>>>> you, lol. >>>>> >>>>> >>>>> On Fri, Mar 13, 2015 at 3:38 PM, Brian Meade <bmead...@vt.edu> wrote: >>>>> >>>>>> 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. >>>>>> >>>>>> On Fri, Mar 13, 2015 at 3:30 PM, Barry Howser <bhowser5...@gmail.com> >>>>>> wrote: >>>>>> >>>>>>> Hello everyone. >>>>>>> >>>>>>> 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). >>>>>>> >>>>>>> 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. >>>>>>> >>>>>>> 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. >>>>>>> >>>>>>> What am I doing wrong? >>>>>>> >>>>>>> thanks >>>>>>> >>>>>>> _______________________________________________ >>>>>>> cisco-voip mailing list >>>>>>> cisco-voip@puck.nether.net >>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip >>>>>>> >>>>>>> >>>>>> >>>>> >>>> >>> >> >
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