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

[--[ UxBoD ]--]

----- Original Message -----
> 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.

Grabbing a SIP debug I see:

<--- Transmitting (no NAT) to 10.172.120.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.172.120.5:5060;branch=z9hG4bK-dadp6piblhin;received=10.172.120.5;rport=5060
From: "User A" <sip:usera at 172.30.14.8>;tag=c3zqlidz1u
To: <sip:userb at seconddomain.com>
Call-ID: 66b3314cc6d1-jxu0nhluv4zt
CSeq: 2 INVITE
Server: secret
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:userb at 172.30.14.8>
Content-Length: 0

And am guessing that as the source from IP matches the Contact: address Asterisk sees that as a loop ?
-- 
Thanks, Phil