As evidenced by various DTMF interop issues (with RFC2833, inband, etc) over 
the years, I would avoid it if at all possible.  

What does it particularly do that can not accomplished by using RFC 2833 or 
(less ideal) inband DTMF?  Or are you attempting to use it as a band-aid to 
address some sort of interop issue with the carrier involved that is wrecking 
havoc with your particular application?

-metik
  ----- Original Message ----- 
  From: Tihomir Culjaga 
  To: freeswitch-users@lists.freeswitch.org 
  Sent: Tuesday, October 13, 2009 3:24 PM
  Subject: Re: [Freeswitch-users] SIP Overlap support?


  i never found it working properly... i always had some interoperability 
issues and i finished having a "dialplan" on my phones being delivered through 
a config file via tftp or http .. depending of the phone capability.

  BTW: using overlap can lead to a greater system load... be careful when 
setting the minimum number of digits you will send in 1st message. I wish you 
luck...

  T.


  On Tue, Oct 13, 2009 at 9:03 PM, Metik <freeswitch-users-l...@metik.com> 
wrote:

    Both support it.  In the Grandstream, I believe it is called Early Dial 
(vs. SNOM's Overlap Dialing).  It can be problematic if you have a device 
somewhere in the middle that doesn't support 484s.

    -metik
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users@lists.freeswitch.org 
      Sent: Tuesday, October 13, 2009 2:01 PM
      Subject: Re: [Freeswitch-users] SIP Overlap support?


      i do think some softphone can do it but i forgot which one it was either 
snom or grandstream



      On Tue, Oct 13, 2009 at 12:12 PM, Tihomir Culjaga <tculj...@gmail.com> 
wrote:

        you need a softswitch.... i'm afraid a SIP phone is not designed for 
overlap...

        T. 



        On Tue, Oct 13, 2009 at 5:26 PM, Dennis <oderm...@googlemail.com> wrote:

          how could we try? we played arround with a snom phone (snom seems to
          support something in this direction, but are not shure, how we can
          test it and how we can see if it is supported or not.

          any hint?


          2009/10/13 Anthony Minessale <anthony.miness...@gmail.com>:

          > have you tried it?
          > I *think* either we did support it or we would with a small patch 
to sofia
          > lib that I cannot recall if we ever got committed.
          >
          >
          > On Tue, Oct 13, 2009 at 8:51 AM, Dennis <oderm...@googlemail.com> 
wrote:
          >>
          >> hi there,
          >>
          >> i would like to ask, if fs has support for something like "SIP 
Overlap"?
          >>
          >> instead of receiving the phonenumber from our carrier in a block, 
we
          >> want to receive the phonenumber digit-by-digit and we want to tell 
fs
          >> when the number is complete. our carrier could send us the 
phonenumber
          >> digit-by-digit, but what about the fs-side?
          >>
          >>
          >> thanks and kind regards
          >> dennis
          >>
          >> _______________________________________________
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          >> 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
          >> http://www.freeswitch.org
          >
          >
          >
          > --
          > Anthony Minessale II
          >
          > FreeSWITCH http://www.freeswitch.org/
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          > Twitter: http://twitter.com/FreeSWITCH_wire
          >
          > AIM: anthm
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          >
          > FreeSWITCH Developer Conference
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          > iax:gu...@conference.freeswitch.org/888
          > googletalk:conf+...@conference.freeswitch.org
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          >
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      -- 
      Anthony Minessale II

      FreeSWITCH http://www.freeswitch.org/
      ClueCon http://www.cluecon.com/
      Twitter: http://twitter.com/FreeSWITCH_wire

      AIM: anthm
      MSN:anthony_miness...@hotmail.com
      GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
      IRC: irc.freenode.net #freeswitch

      FreeSWITCH Developer Conference
      sip:8...@conference.freeswitch.org
      iax:gu...@conference.freeswitch.org/888
      googletalk:conf+...@conference.freeswitch.org
      pstn:213-799-1400



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