Genesys CTI User Forum
Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: justany1 on August 23, 2009, 10:25:00 AM
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Hi There,
Was searching this forum but no luck.
Am trying to implement SIP Server version 7.6 in a Genesys Express environment to work with a AudioCodes Mediant 1000 with E1 interface incoming.
In order for calls to come to agents & for agents to be able to call to external, we’ve set in CME Switch DN, a DN with type trunk, & configure inside IP address & port number of the AudioCodes, call can successfully come in & out.
But in order for the incoming call to target a group, do I need to setup a Routing Point in CME Switch DN, so we can apply Routing Strategy to it?
Any guru can share some light?
Thank a million.
Regards
any1
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Hi,
Yes, you're right that Routing Point DN is required if you want route the calls using Genesys routing. Routing Point number has to correspond to number defined as destination number on Audiocodes gateway (section Routing Tables / Tel to IP Routing). If you are not sure what numbers are configured there you can check INVITE message in SIP Server. Routing Point DN number have to match user part of Request URI or user part of "To" header if Request URI is empty.
Some information related to integration of Audiocodes GW with SIP Server can be found the "Framework 7.6 SIP Server Integration Reference Manual".
R.
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Yap, it work like a champ....
Thanks René, you're the best...
Regards
justany1