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

[Kyle Kienapfel]

On Tue, Jul 6, 2010 at 11:08 AM, --[ UxBoD ]-- <uxbod at splatnix.net> wrote:
> ----- Original Message -----
>> ----- Original Message -----
>> > 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.
>> >
>>
>> From what I have read "hairpin" calls are not supported by asterisk;
>> so am guessing something has been fixed in the 1.6.2.X branch that
>> should have not worked in 1.6.1.X anyway :) While I continue the
>> research have implemented using a workaround via the AstDB and the
>> following changes to the uri-dial plan:
>>
>> exten =>
>> _[a-zA-Z0-9].,n,GotoIf(${DB_EXISTS(URI/${EXTEN}@${SIPDOMAIN})}?inturi:exturi)
>> exten =>
>> _[a-zA-Z0-9].,n(inturi),Goto(${DB(URI/${EXTEN}@${SIPDOMAIN})})
>> exten => _[a-zA-Z0-9].,n(exturi),Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>
>> This is a bit of pain as we have to make sure we update the DB when a
>> new inbound URI is added; though it works and means we can stick with
>> the 1.6.2.X branch.
>>
>> Would be interested to hear from a dev though as to whether they think
>> it should work as we originally had it configured ?
>
> Do you think this should be raised as a issue in bugtraq or at least brought up on the asterisk-dev mailing list ?
> --
> Thanks, Phil
>
> --
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Turn on sip debug for everything, posting just one sip packet doesn't
tell much.
Knowing if asterisk is sending udp packets to itself or not is a
fairly important detail.

I'd go with the issue tracker