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Baze

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Posts posted by Baze

  1. 37 minutes ago, Jim_PGi said:

    @baze is it possible to change the recording that plays for a forwarded call on a per-call basis?

    Well technically anything is possible I suppose?  You'd have to change the recording file itself which I've not done before.  But once it is changed it should just pull that file instead of the existing one.

  2. Kind of confused - you are using broadsoft as the back end phone system?  This forum is for 2600hz (a competitor back end) although who knows, maybe they are working on something I'm not aware of.  I can certainly tell you how to do that with the 2600hz backend, but no clue how to use broadsoft.

  3. TBH in our testing lawyers (again to the high maintenance thing) seem to prefer the normal G711 sound.  Any change to that is a problem - we even once had to tweak sidetone settings on the VVX 500 series because they couldn't "hear themselves" talking in the handset.  Opus is great, but ultimately for compatibility we settled on plain old G722 for clients that don't mind the HD sound.

  4. Hey Jack - the router definitely wouldn't affect things.  There are a number of things I know we've seen here, but specifically you might look at your phone configs and see if you can increase the "handset volume" which makes the phones transmit a louder stream.  Pretty much every model of phone will have that.  Obviously the speaker or handset volume on the receiving end should be checked to see if they have their volume up.  If it keeps resetting to a medium volume, there are settings usually to allow for "fixing" the default volume to be higher or to remember the last setting.  

    Here is the setting we use to set the microphone gain (volume) of the phones for Polycom:

    voice.gain.tx.digital.handset

    Here are our Polycom settings to make sure the volume settings remember the last settings:

        voice.volume.persist.handset="1"
        voice.volume.persist.headset="1"
        voice.volume.persist.handsfree="1"
        voice.volume.persist.usb.handsfree="1"
        voice.volume.persist.usbHeadset="1"

  5. Hey All - wondering if anyone has come across this one yet.  By default, 2600 is set for **+ext to be the "call direct to voicemail" option.  On older firmware versions of Yealink this worked fine.  After updating to 35.83.0.50, the dial pad now does an IME style thing where it cycles through, so if you hit ** you end up getting a . instead.  We've tried modifying the ime.text (language) file to no avail.  Got a ticket in with Yealink because at the very least this seems poorly executed or at least should have a config option.  Not sure what firmware version the hosted platform runs either.  

    Anyone run across this or have a fix not involving downgrading firmware or manually changing that star code on all clients?  Thanks in advance!

  6. Yep!  that is the one.  If they aren't using smartpbx, you don't have much of a downfall.  The main issue with modifying those callflows is they can be "regenerated", but looks like at least for the 0 it won't affect things like it did.  So in your case, you should be able to remove from there, save, and then go in and create a new callflow using 0 and have it do whatever you need.  :)

  7. ^true - just keep in mind the user has to be logged in as super duper because you will have to remove the 0 from the main callflow first which is an automatic one.  Otherwise you will get a "must use unique" error when building the new callflow.  It does look like smartpbx will rebuild without giving an error though which is great!  Before if you tried to hijack 0 to a new advanced callflow you'd get an error when you went back to smartpbx.  

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