Setup Asterisk (FreePBX) with Obi to use the AA features of the Obi
obiliving:
QBZappy,
It seems there is a conflict on what you want, or I have misunderstood your intention.
You want one particular unknown CID caller to go to aa, and yet other callers to go the ph?
By unknown CID I assume you mean it has a caller-id number, just not known to you.
The problem I have is how to tell which caller to go to aa and which caller to go the ph, since they are all unspecified.
If you truly want any caller number to go to aa, then just specify {aa}
in the InboundCallRoute (i.e., everyone is in your circle of trust), and don't include
{ph} in the route. Will they work for you?
Sorry if I misunderstood your intention. Please clarify.
ShermanObi:
Would this work? ...
Use the following for ITSP InboundCallRoute={aa},{ph}
The OBi Attendant would pick up all the calls (after the AA answer delay timeout) and the payphone caller can then access the AA menu and make a new call (option 2) to an Asterisk extension, OBi No. (**9 or speed dial). You would just need to be careful here because all inbound calls would get the AA and unless there is a PIN, everyone could use option 2 or 3 to make a new call.
Note sure I understand the part about listening in for the tones and then switching to AA.
QBZappy:
Obiliving,
That's just it. Obi AA IVR must screen all the calls in order to precess them.
I could make this work if:
1) I can figure out how to call forward using the "Follow me" feature of Asterisk and pass an extension number to the call forwarded extension. The reason for this is that the first call will show CID of the telephone number calling. The first extension dialed can be "Forwarded", hopefully passing the extension number. Can someone help me with this part?
Ex: Call Asterisk, dial ext 703, this will be forwarded unconditionally to ext 704, which can be hard coded in the ITSP InboundCallRoute to go to AA.
or
2) If the Obi could be configured to answer all calls silently we could have it AA all calls. The admin of the unit could instruct users what to dial in if they wanted to dial Option 1,2,3 during this silent interval. Short pause then if no entry goto Option 1 to continue this call. Timing of these pauses would have to give impression that call flow is not interrupted. I don't suppose this unit will ever have the ability to upload a recorded message to control the IVR. This could introduce the impression of a nested IVR feature if Asterisk was in front controlling the first part of the call process.
I remember when answering machines were common. I had to setup a fax machine, telephone, and answering machine in a small office with only one POTS line. Answering machine picked up first. I had to make certain there was silence in the beginning of the message so that the fax machine could intercept an incoming fax. If not a fax then call would then be picked up by someone or go to the answering machine.
In any event whether Asterisk is in the middle or Obi is stand alone, this DISA feature could be implemented. The unit already has DISA in its current state. Option#2, and callback Option#3 as a bonus. I haven't seen any other ATA with this feature.
There is always the solution of setting up a dedicated Obi to AA all calls. Takes away flexibility of the unit.
ShermanObi,
Your suggestion might be the way to go. Wanted to setup a more seamless call flow without listening to Obi IVR on every call.
Tks for your feedback.
MichiganTelephone:
Regarding #1, maybe this would help you — it's an unsupported third-party module for FreePBX:
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/set-callerid
Quote
Set CallerID
Adds the ability to change the CallerID within a call flow.
Set CallerID allows you to change the caller id of the call and then continue on to the desired destination. For example, you may want to change the caller id from "John Doe" to "Sales: John Doe". Please note, the text you enter is what the callerid is changed to. To append to the current callerid, use the proper asterisk variables, such as "${CALLERID(name)}" for the currently set callerid name and "${CALLERID(num)}" for the currently set callerid number.
The latest release can be found at:
http://www.freepbx.org/trac/browser/contributed_modules/release
Be sure to get the most recent version. So basically what you do is install this module, create a CallerID instance where you force the CallerID name and number to be whatever you want, and then in your inbound route you can direct calls to your Caller ID instance (it should appear in the destination list). If you need to access it from someplace in the call flow that doesn't support destinations directly (such as Follow-Me) I suppose you could assign the CallerID instance an "extension" number using Misc. Applications (it says you can assign "Feature Codes" but a feature code can be any number, it does not have to be a *xx code or something like that). So your call flow might look like:
Inbound route --> CallerID instance --> OBi110 Extension
or perhaps
Inbound Route --> IVR --> Selected Caller ID instance --> OBi110 Extension
The latter would let you use a FreePBX IVR which could then be used to select one out of several Caller ID instances, each of which would send a particular (but different) CallerID to the OBi110 (which the OBi110 could use for additional processing).
I think, if I am understanding you, this would eliminate the need to try and set up a fake extension.
If you've never downloaded a third-party module before, the trick is to click on the first .tgz link you see (don't right-click yet) and it will take you to another page (example: http://www.freepbx.org/trac/browser/contributed_modules/release/setcid-2.8.2.tgz ). On that page you will see a link that says "Download in other formats: Original Format." Right click on the "Original Format" link and download the file to your system. Edit the filename so that it ends in .tgz (strip off the "?format=raw") and then go into Module Admin in FreePBX and click on "Upload Module" and go from there.
QBZappy:
Just a follow up.
Managed to pass CID from one extension to another using FreePBX. In the FreePBX Follow me setting for an extension there is a setting for "CID Name Prefix:". When we dial an extension which forwards to another extention internally the CID Name Prefix: we use is passed on the receiving caller ID device. However AA did not process the call. I made a trace of the call and noticed the SIP header did not have the prefix.
I'm giving up.
Would have been interesting to have something I have come to call my "Stealth circle of friends" feature on the Obi.
Still the best ATA I've ever bought. I recommend this one.
Note that ShermanObi has setup a how to for automatic callback using the "ITSP InboundCallRoute". You can also add extra processing to the inbound calling. In addition to his example I added the following underlined section:
{(<**2>(5145551212|5145551234)):aa($1)},{102|112|703|704|):aa},{ph}
------------------------------
Which adds a few choice internal extensions from an Asterisk box and a peered Grandstream 5024 PBX into the mix, giving them the ability to Google voice out some toll free calls. Calls from 5145551212, and 5145551234 calling in must and hang up before the AA picks up if you want a callback on the CID which made the call. The callback offers Options 1,2,3. It's one short ring before AA picks up. If you don't hangup you are offered the AA with the 3 Options giving you another chance to choose option 3 for you to dial the number to call you back.
The second part after the comma offers internal extensions 102,112,703,704 the AA offering the 3 Options, all other calls ring the attached phone (phone port).
This callback feature is reason enough to use this product for the cell phone users who have unlimited incoming air time.
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