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

[Kyle Kienapfel]

On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- <uxbod at splatnix.net> wrote:
>
> ----- 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 ?

I don't know these things, but you should probably post more of a SIP
trace. Maybe turn on full sip debug to a file for long enough to see
what the SIP conversation looks like that asterisk 1.6.2.9 is having
with itself.