[Asterisk-Users] 482 Loop Detected when transferring calls back to Asterisk

[David Allen]

Hi,

I want to be able to receive incoming calls via H323 to Asterisk for SIP
Conversion and then send the Call to a seperate machine running SER to route
the call to the end user CPE. However if the call is not answered, I want to
be able to send that call back to the machine with Asterisk on it to a
Voicemail Box. I have set it up this way, however I'm getting a Loopback
Detected each time the call is sent back to the Asterisk Box, which then
tries to connect the call as a fall back to a Local Channel listed in my
default context.

The Call Flow is as follows:

-------->Asterisk (Performs H323 to SIP Conversion)
----------------------->Passes the call to SER ------------------------->
UA/CPE is called
                                                   |
|
                                                   |
|
                                                    ---------------------
On timeout the call is sent back to the Asterisk Box --------------
                                                                             to
a Voicemail Box on that machine.

Is there anyway around the loopback issue on the Asterisk Box (either by
using another IP Address or by some how mangling the SIP Message)?

Thanks,
David
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