[Freeswitch-users] SIP Overlap support?

[Tihomir Culjaga]

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-list at 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 <anthony.minessale at gmail.com>
> *To:* freeswitch-users at 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 <tculjaga at 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 <odermann at 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.minessale at 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 <odermann at 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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>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
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>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> > pstn:213-799-1400
>>> >
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>>> >
>>>
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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_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
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