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
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gsm:   06 66 98 76 34

Service Client IPeva
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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
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