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 ________________________________ DISCLAIMER: This email may contain confidential information and is intended only for the use of the specific individual(s) to which it is addressed. If you are not the intended recipient of this email, you are hereby notified that any unauthorized use, dissemination or copying of this email or the information contained in it or attached to it is strictly prohibited. If you received this message in error, please immediately notify the sender at Cyient and delete the original message.
