[Freeswitch-users] rtp-autoflush

[Mouncif Benniane]

One more question, will it participate in codec negotiation sent by the
inbound voip provider?
Right now I have it setup this way:

Incoming Inbound DID (voip provider --> freeswitch(media bypass mode) ->
endpoint sip device.

Thanks again!

On Wed, Jan 13, 2010 at 6:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its better to use one of the other but it probably is ok
>
>
> On Wed, Jan 13, 2010 at 5:20 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>
>> thank you, also is it okay to set bypass_media_after_bridge=true and
>> bypass_media=true at the same time?
>>
>>
>> On Wed, Jan 13, 2010 at 6:08 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> no, as the name implies bypass_media does not touch media whatsoever and
>>> hense rtp related settings do not come into play.
>>>
>>>
>>> On Wed, Jan 13, 2010 at 5:00 PM, Mouncif Benniane <mouncifbb at gmail.com>wrote:
>>>
>>>> We are running freeswitch with bypass_media=true, if we change the
>>>> following settings :
>>>>  rtp-autoflush-during-bridge and  rtp-autoflush
>>>> does it affect anything?
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
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>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
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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:+19193869900
>
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