[Asterisk-Dev] Incorrect behavior : address for SIP responses
Christian.Cayeux at alcatel.fr
Wed Jun 8 23:49:49 MST 2005
I've tested again but 'nat=never' doesn't change anything and there's no
rport : * defaults to RFC3581.
I've read the code again, and on my understanding, * actually defaults to
As you suggests, i'm going to open an issue.
De : asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]De la part de Kevin P.
Envoyé : mercredi 8 juin 2005 17:57
À : Asterisk Developers Mailing List
Objet : Re: [Asterisk-Dev] Incorrect behavior : address for SIP
Christian Cayeux wrote:
> So, as Asterisk is intended to support RFC3581, should we consider it a
> Well, i still believe this is a bug, because this is the cause of
> serious interop issues. If * is intended to interop with most of SIP
> implementations, then it should conform to RFC3261.
Asterisk defaults to RFC3581 processing, unless you specify 'nat=never'
in your configuration. However, if the SIP peer is not specifying
'rport' in their headers, we shouldn't be using RFC3581-compliant replies.
If you know that this is the case, please open a bug in Mantis with a
_complete trace_ of the call. If you don't know how to generate one, get
on #asterisk on freenode and ask a bug marshal to help you get one made.
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