[asterisk-users] SIP response 482 "Loop Detected"

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Hi,

We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same.

Any ideas on how to overcome this problem as we dial our clients based on their email address.
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Thanks, Phil