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). 2010/10/19 David Ponzone <david.ponzone@ipeva.fr>

Nicolas,

you may try to set sip_via_rport in the dialplan, but I don't know if this will work. I suppose it depends on what you are trying to defeat.

David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34

Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com

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Le 19/10/2010 à 14:51, Nicolas Brenner a écrit :

Hello, I need to either remove the rport parameter from the Via header on an initial invite or set its value to a specific port. I've looked through the mailing archives and the wiki but I haven't found anything. I appreciate any help or pointers I can get, thanks.

Nico _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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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

Thanks for all the help. The provider is a cellphone operator in my country, they usually don't interconnect with anybody (at least not through SIP), but are giving us access to this only because they want the calls from our service to be routed directly to them. Unfortunately I have no access to the router, we rent this server on The Planet, but we have never had any problems like this with any other provider, so I'm guessing this doesn't have to do with nat. Anyway, I was finally able to solve this by setting transport to tcp instead of udp on the gateway config with: <param name="register-transport" value="tcp"/> Even though it's working fine now, I'm not that happy, because I have no idea why it didn't work with udp. Well, hopefully the provider/customer will be glad this got solved. Again thanks to everyone for all the help. I'd be happy to answer any questions if you are curious about this or have a similar problem. Best, Nico On Tue, Oct 19, 2010 at 11:45 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:

If the top of their standards are to make it work with asterisk, they are clearly not very wise.

Are you using some kind of asymmetric nat? Why is it getting a random port? Even with rport enabled it should be the true port the packet is seen coming from and a return path should be expected the same way.

Check your router for asymmetric nat or SIP ALG features that you can disable in it's config.