Ring Group separation

[Clivusmul]

Newbie here... I have a FreePBX Linux setup with multiple extensions and two incoming routes from Flowroute.  Just installed a new OBI-202 to hopefully improve CID shown on a Dish connected TV..  Everything seems to work great including FAXing however, the Obi-202 RINGS both analog ports whenever a call is routed to either ATA port by the PBX.  Substituting my prior Cisco ATA shows proper signaling of the analog ports.  A call to "line-1" rings the ring group associated with "line-1"; calls to "line-2" ring the proper lines... In the OBI-202, I have two (2) SIP setups with two distinct extensions (113 & 114) and both are properly registered with the Linux PBX machine.  The ring groups in the PBX are properly configured.  My guess at this time is that something is mis-configured in the OBI-202 causing both SIP lines to ring independent of which line was called; it is if they are slaved together... Suggestions on how to configure the OBI-202 to ring the analog phone ports independently/properly would be greatly appreciated.  Thanks, BobJ (Clivusmul)

[azrobert]

Check the following in OBi Expert:

Voice Services -> SP1 Service -> X_InboundCallRoute: ph

Voice Services -> SP2 Service -> X_InboundCallRoute: ph2

One or both are probably set to: ph,ph2

[Clivusmul]

Thanks for the very prompt reply!  Yes, the are both set to "ph,ph2".  I tried changing SP1 to "ph, ph1" but that did not alter the behavior.  The problem with this anomoly/mis-configuration is Line-1 is my voice analog line to cordless home phones and "Line-2" is my office FAX line.  The fax line is supposed to only ring to the fax machine and it is set to auto answer on the 2nd / 3rd ring.  If no one is home and a voice caller rings the house, after 3 rings the fax attempts to answer and there is no way for a caller to leave a message... Might be a great setup to discourage spam callers  

I will try the suggested settings and report!

BobJ

[azrobert]

ph is short for ph1 and both point to Phone Port 1.

[Clivusmul]

Thanks for the great prompt/configure advice... Now the ringing works properly... Next I need to know how to access messages from the analog phone with the blinking light... The old "*" code for dial voice mail in the FreePBX box seems to be directed to another use in Obi-202... I'll work on this for awhile!  Thank you again,
Bobj

[azrobert]

If you want to send * to SP1 and SP1 is your primary route, just add *| to the ITSP A DigitMap.

[Clivusmul]

AZRobert, Here is my current default ISTP A digit map -

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.)

Can you suggest the exact syntax for the command line addition?  Do I simply add "*|" inside the first and last () after the "." at the end of the string?  My Trixbox/FreePBX has a voice mail feature code of *98.. I can set it to anything however... *97 is unused in Obi202 currently and it can be used I think...  I can easily change it to *97.  Am I on the right track?  bj

[Clivusmul]

AzRobert, I copied the suggestion given in the forum for dial plan for Trixbox/FreePBx as follows and it works fine! 
(911S0|1xxxxxxxxxxS0|011xx.S2|xx.S2|xS2|xxS2|*X.S2|(Mipd)|[^*#]@@.)

Really appreciate your help!  bj

[azrobert]

First, let me explain a few things about the DigitMap. You should eliminate any unused rules in the DigitMap.

"xx." will match 1 or more digits. If a dialed number gets an exact match in a DigitMap and can't match any other rule it will be immediately routed with no delay. "xx." will match any dialed number, so the OBi will wait for additional digits. The pause can be 3 or 10 seconds.

"(Mipd)" and "[^*#]@@." are used to match an IP address.

<1>[2-9]xxxxxxxxx will add a "1" prefix to a 10 digit number.

Finally here is the DigitMap I suggest:
(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|<1aaa>xxxxxxx|*98)

aaa is your local area code for 7 digit dialing.
You don't have to change the StarCode profile. The DigitMap takes precedence over the StarCode profile.

Edit:
I see you have a solution. Maybe you can combine the 2 DigitMaps. The Sx suffix overrides the standard timer. S2 will wait 2 seconds after a match.

[Clivusmul]

Kind thanks again AZRobert... I will give that a try also... FYI, my FreePBX /asterix box also has dialing rules in and/or associated with Trunks/Outbount Routes, etc... Things like Prepending dialing prefix and so forth...  What I needed to do and have now down is allow Feature code dial sequences made on Analog handsets to pass directly to the PBX so it can route the number per the rules... Without making your initial change, the OBI202  was intercepting the *97 for it's assigned purpose and not allowing it to be ordered by the PBX...

I seem to have it working now quite well and am very impressed with the audio quality... I am only using the Obi202 for ATA purposes... Most of my handsets are IP based and are directly registered with the PBX...  Thanks for passing along the "(Mipd)" and "[^*#]@@." are used to match an IP address" items... I was wondering what they did!  bj

[azrobert]

AzRobert, I copied the suggestion given in the forum for dial plan for Trixbox/FreePBx as follows and it works fine! 
(911S0|1xxxxxxxxxxS0|011xx.S2|xx.S2|xS2|xxS2|*X.S2|(Mipd)|[^*#]@@.)

The above DigitMap might cause you problems by routing the call too early. If you have a phone where you dial the number then hit Talk/Send then you won't have a problem. If your phone is taken off-hook when picked up, you might have a problem. The rules with a S2 suffix will cause a call to be routed if you pause for 2 seconds. Rule "xx.S2" will route the call early if you dial any number of digits and pause for 2 seconds. Even the DigitMap I suggested will route a 10 or 11 digit number early if you pause for 3 seconds after dialing 7 digits.