Inbound route with destination set to Call Flow Control can't be transferred correctly

I have created a call flow control for “Lunch Time” so incoming calls through an inbound route can ring on the dinning room when activated.

We are using Grandstream GXP-2110 phones with auto-attended transfer enabled and BLF keys for the extensions.

If the inbound route destination is set to a sip extension, I can correctly use BLF keys to transfer the call pressing TRANSFER button, then BLF key, announce the call, then TRANSFER key again to transfer.

But if the inbound route destination is set to the call flow control, I can’t use that sequence: after pressing TRANSFER key, the BLF button does nothing, so I have to manually dial using the dialpad, anounce the call and press TRANSFER button again.
If I press HOLD instead of TRANSFER, then I can use BLF keys to dial, but that way I’m doing an attended transfer instead of auto-attended, since after dialing destination with BLF key, I have to press TRANSFER button, anounce the call and then the held LINE button instead of TRANSFER button.

Is this a problem with the call flow control? Or is it some sort of incompatibility between my phone’s “auto-attended transfer” function and the call flow control?


Typically the way I have setup a similar system for clients in the past who eat lunch in their conference room, but do not like that phone to be part of a normal ring group, is create another ring group that includes the conference room. Your call flow control would be your normal routing (whatever you have it set up as now), with the override being the other ring group (or destination extension if you just want the one phone to ring when override is on). If you are using ring groups vs IVR, queue, etc normally, you may want to still include those phones as well in the new ring group ETC because some people may be working at their desk thru the lunch hour waiting for a client to call back ETC.

As far as the BLF / speed dial button, you should be able to program one of your line keys on your grandstream to speed dial & BLF *280 or whatever the feature code is to toggle the override.

I used a lot of SPA9xx and SPA5xx series cisco phones, it’s pretty easy to reprogram a line button on those for BLF/speed dial, I’m sure it is pretty easy on grandstream phones as well…check the manual of your phone for instructions.

Hope that helps!

Inbound Route > Call Flow Control “A” > Normal (Your usual queue, IVR, ring group etc) or Override (Your cafeteria room phone/Ring Group includes cafeteria)

Check the dial options on the transfer in both cases. Maybe the options get stripped somewhere.

T: Allow the calling user to transfer the call by hitting the blind xfer keys (features.conf). Does not affect transfers initiated through other methods.

If you have set the variable GOTO_ON_BLINDXFR then the transferrer will be sent to the context|exten|pri (you can use ^ to represent | to avoid escapes), example: SetVar(GOTO_ON_BLINDXFR=woohoo^s^1); works with both t and T 

t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf) Does not affect transfers initiated through other methods.


t: Allow the called user to transfer the call by hitting the blind xfer keys (features.conf) Does not affect transfers initiated through other methods.

Nothing to do with call flow control. I was using two sip accounts on the phone, one for internal calls and another one for external, kind of simulating some sort of ksu, I know… asterisk was not designed for that. Anyway, the phone needs the account to be defined for blf keys to work, so since external accounts come from account “A” but blf keys are set to account “B”, transfering fron external to internal using blf keys doesn’t work. Thank you both guys for your help! Now I’m dealing with some sort of bug where calls established to or from a particular extension (106), the other party keeps hearing the “ringing tone” even after the call has been answered. Will create a new post for that.

Yes, the transfer has to happen on the same line. This is not a limitation is Asterisk, however.
For asterisk, they two lines are separate entities, so it cannot bridge the call elsewhere on the server level.