I hate to ask but what sort of provider uses Avaya SES? For what it's worth I've done interop with Avaya SES and FS before. No problems with rport there... On Tue, Oct 19, 2010 at 10:24 AM, Nicolas Brenner <nico@clickfono.com> wrote:
Didn't work. My problem is the following: I have a VoIP provider that when it gets an INVITE from FreeSWITCH, instead of replying to the originating port (5080), it replies to a random port, and of course FreeSWITCH doesn't get the reply, so it keeps sending the same INVITE until it times out, even though the call is actually made and the call status is reported by the provider, although to some random port where nothing is listening. I talked to the provider and they are not going to change any config, but told me it works fine with Asterisk and told me to "disable rport and nat". Only there's no explicit option to disable nat (like nat=no on Asterisk's sip.conf) nor to remove the rport parameter from the Via header (which I'm trying to figure out if it would solve the problem and make the provider's SIP server reply to the originating port).
-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com