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.

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*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

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*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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