PJSIP / SIP chans INVITE source port flapping

Hello Guys.

Im having a trouble that I can’t got to solve. I’m doing a interconnection between 2 machines, one Asterisk from my side and a IMS of a Carrier.

Is mandatory for that carrier, that the request comes from port 5060.

I used SIP channel with Asterisk 21 and PJSIP channel on latest certified asterisk. On SIP CHAN (Ast21) all my invites comes from random port 45784, for example. With PJSIP on latest certified, my invites come randomly from 5060 or randomized ports.

Is there any way to force my SOURCE PORT to use 5060 to 5060 always? Im not natting, is a direct connection between a BGP.

Thanks in advance.

Certified version of Asterisk are not supported by the open source community. They are only intended to be supported through a support contract with Sangoma, and may be missing features that are not covered by such contracts.

Are you using TCP? For UDP, at least, the port should always be the one to which the transport is bound.

I am more than sure that your asterisk does always use the same source port.

I guess you are behind a NAT firewall and it chooses random high ports as source port when routing your pakets to the internet. It’s more or less normal behavior in NAT scenarios and in fact I also had providers from time to time who would complain about that.

If my guess is right, you should check on your firewall. Some let you set rules so that your pakets always go out with the same source port.

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