[cisco-voip] SIP Anti-Trombone

[Bob Zanett (US)]

Steve,
With RNA, the call has not been setup and so should roll to Unity just fine.   If you have any issues, place router-based software MTPs on your Cube gateway.  Many times you will encounter issues with transfers, conference, etc with SIP users over a SIP trunk.  The reason is that the re-invite is sent to the telco and many times they do not handle those.  By tying to an MTP, you keep all call manipulation in-house and at the place of origin into your enterprise.
I have done this before, but I am more than willing to setup a WebEx or Live Meeting and go over SIP Trunking and some of the ins and outs.   If  you are interested, please send me a private email.
Cheers,
Bob
Dimension Data
www.dimensiondata.com

From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of STEVEN CASPER
Sent: Friday, September 24, 2010 8:05 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] SIP Anti-Trombone

If I make an inbound call via a SIP trunk to a remote site will the call use two paths (thus using double the WAN bandwidth) if the call RNAs to Unity Connection? For instance will it do this:

SIP Trunk------CUBE----WAN------Remote Site
                                                      |
                      Unity----WAN----------|

or if there is an Anti-Trombone feature the call would end up like this:

SIP Trunk------CUBE-----Unity


Steve

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