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FMFM FIND ME FOLLOW ME

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uniquename4me

Technical User
Oct 31, 2013
184
CA
I've just been testing FMFM and wondered whether anyone can comment on possible solutions to some problems I've encountered?
1. If one active call comes into a phone and sent through FMFM, it won't process any additional through the same extension?
2. If the call comes in as "anonymous", since the calls are being routed back out on SIP lines, the provider won't process calls without caller ID. As suggested by a previous thread, putting public OLI on the virtual DN's does not work as it still attempts to redirect the caller ID from an incoming caller. Putting caller ID on the SIP account will work, but it is a lot nicer to have the source caller ID passed through when answering random calls outside of the office.

Ideas/Solutions?
 
it won't process any additional through the same extension?"

Not sure what that means, can you rephrase or detail?



I am not so sure I get what your saying on the second issue also, sorry but this one seems confusing.
You have no CLID coming in therefore your FMFM does not call out and is rejected? is that what you mean?


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Ok after reading the second one ten times...

FMFM indeed works but will not process Anonymous calls because you removed the CLID from the account.

Most accounts indeed need the CLID (your phone number or account ID) in the account setting an/or the OLI under sets.

In IP trunks/General/Settings/ change the Forward OLI to "Last Redirect"

I have OLI/Set and just the PAI CLID Override in accounts as well "Last Redirect" and the original callers CLID will show on my mobile instead of my DID.

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1. If one active call comes into a phone and sent through FMFM, it won't process any additional through the same extension?"

Do you mean 2 simultaneous calls will not FMFM the last one in?

I have never had this issue but things I would look at:
Check to see how many channels your provider allows you to use to call in or out.
Check how many Virtual Terminals you have
Check how many sessions you allowed yourself in Callpilot.
Check how many Intercom keys you allowed the Applications DN's to have.


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Ok.

Thanks Curlycord!

I tried your settings and by putting the phone number in the PAI CLID Override and setting last redirect, the phone number shows up on all calls both dialed through the FMFM as well as dialed from a set within the premises. So it seems to override all outbound CLID. As for "anonymous" calls, they still won't go through on FMFM just ring on the internal set.
The simultaneous sessions are set to 12, I set the virtual extension to 5, and I've put multiple line appearance as well as multiple IC links with no apparent difference? When I call with a second call, whether the first has been answered by the cell or not, the second call is not re-routed by FMFM out to the cell?
 
Curlycord.

Found the multiple calls under the Class of Service in the CallPilot, so I'm testing it now.
 
Ok Curlycord!

I managed to change a bunch of things and get everything working ideally! The only remaining thing that I can't seem to figure out is that although the it will now present multiple calls (2) I have it set to 3, but can't get the 3rd call to go out? It's more a curiousity at this point to see where the fault lies.
 
Well that's good news for CLID.

What about channels? how many are provided by your carrier?
Does it even try when you look at Monitor?

Just to be clear the Virtual terminals would be Application DN's and while you make sure you have 5 or so Intercoms also make sure all of them also have BlocA Line Pool Access



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The virtual DN's all had 5 IC appearances just to make sure that there would be no issues and they all of course have line pool access to BlocA but the limit seems to be 2 calls presented so far. The SIP lines as per my provider are supposed to be open to a lot more calls than the 2 being processed.
 
I checked the active call limit in the accounts section to see what it was set to and it was left as default which was 0 figuring that if you didn't set one, it was as determined by the carrier. I figured that likely the only time you'd set one was on a BCM50 with the 8 SIP trunk limitation? It didn't seem to have a bearing given that 2 calls went out on the FMFM? With a 450, I'm not sure what I'd restrict it to?
 
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