[cisco-voip] ATA190 Plar SIP Dial Rule

[Barry Howser]

[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 <bmeade90 at 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 <bhowser5050 at 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 <bmeade90 at 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 <bmeade90 at 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 <bhowser5050 at 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 <bmeade90 at 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 <bhowser5050 at 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 at puck.nether.net
>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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