[cisco-voip] ATA190 Plar SIP Dial Rule

Brian Meade bmeade90 at vt.edu
Fri Mar 13 17:25:16 EDT 2015


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
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20150313/e670b124/attachment.html>


More information about the cisco-voip mailing list