Anyone found a solution to this problem?
Remi
Damon Estep wrote:
I have considered opening a bug report on this, but wanted to get some
feedback and make sure I am not missing something in the way of a
simple work around. What is the scenario in which this impacts your
implementation?
Ours is the desire to use the same realtime SIP database for many
asterisk servers, and route the call based on a “home server” value in
the realtime database. The problem is that a call routed form one
server to another will not complete because the originating server is
not trusted as it should be by IP address, rather the SIP UA that
initiated the call is expected to authenticate on the destination
server, which is ridiculous.
All methods of allowing un-authenticated SIP peering (host=,
insecure=) are broken as soon as the caller name portion of the “from”
header URI is present on the called parties server.
I can not think of why it would break something different to reverse
the evaluation order.
------------------------------------------------------------------------
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Doug
Meredith
*Sent:* Thursday, January 04, 2007 10:23 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] asterisk sip peer/user matching
methodsforauthentication backwards?
Hi,
I too have found this matching to be frustrating. I would like it to
behave as you describe.
Doug
--
Doug Meredith
506-854-7997 ext. 801
------------------------------------------------------------------------
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Damon
Estep
*Sent:* Thursday, January 04, 2007 1:50 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] asterisk sip peer/user matching methods
forauthentication backwards?
Take an example where there is two sip users defined in sip.conf as
follows;
[peer1]
Host=192.168.1.1
…
[peer2]
Host=dynamic
Secret=password
…
[Peer3]
Config not relevant
…
The intention is to accept calls from peer1 without authentication (ip
address authentication only), but require authentication from peer2
If by chance a SIP invite comes “From” [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> (where the name peer2 on the calling server
coincidentally matches a defined sip user on the called asterisk
server) “To” [EMAIL PROTECTED], Asterisk will attempt to
authenticate the caller “peer2” rather than accepting the call based
on the fact that it came from a trusted Ip address defined for peer1.
Since peer1 is trusted it is not sending credentials and will have its
invite rejected with a 407 “proxy authentication required” when it
fails to authenticate as “peer2”.
This logic seems backwards to me, the IP address should be matched
first, and if there is no statically defined user with that IP address
the username should be matched next. This would insure that all calls
from the trusted IP address are accepted regardless of whether there
is coincidently a SIP user with a matching name defined on the target
asterisk server.
So rather than looking for a match in this order;
1. name portion of “From” URI in the invite (“host” in the URI
[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>).
2. ip address statically assigne for a user
it should look in this order;
1. statically defined sip user ip addresses
2. name portion of the “From” URI
Can anyone shed any light on this, or suggest a workaround so 407’s
are not sent if the invite “from” header happens to have the same name
portion of the URI as a defined sip user on the target asterisk server ?
------------------------------------------------------------------------
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