Actually it being ued to create structure like (phone + "\" <sip:" + phone
+ "@" + proxy)
not sure why

On Tue, Sep 29, 2015 at 11:54 PM, Jeff Clay <[email protected]> wrote:

> Reading http://openmeetings.apache.org/red5sip-integration_3.0.html
>
>
>
> Insert proper values to the /opt/red5sip/settings.properties
>
>
>
> red5.host=127.0.0.1 # red5 server address
>
> om.context=openmeetings # Openmeetings context
>
> red5.codec=asao
>
> red5.codec.rate=22 # should correlate with mic settings in
> public/config.xml
>
> sip.obproxy=127.0.0.1 # asterisk adderss
>
> sip.phone=red5sip_user # sip phone number
>
> sip.authid=red5sip_user # sip auth id
>
> sip.secret=12345 # sip password
>
> sip.realm=asterisk # sip realm
>
> sip.proxy=127.0.0.1 # address of sip proxy
>
> rooms.forceStart=no # TBD
>
> rooms=1 # TBD (not in use)
>
>
>
>
>
> What is the actual purpose of the sip.phone parameter? DID’s and
> extensions (aka phone numbers) aren’t related for the purposes of
> authentication a sip user in asterisk.
>
>
>
>
>
>
>
> *Jeff Clay*
> Network Administrator | Information Technology
> *Cyient*
>
> *Direct:+*1 870 215-5506 | *Board:*+1 870-236-1080 ext 1506
>
>
>
>
>
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-- 
WBR
Maxim aka solomax

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