Using Asterisk as SIP relay server

2
votes

I currently have issues with SIP User Agents behind a symmetric NAT connecting to my SIP client, which is an IVR voice service. I read that Asterisk has a solution for this in the sip.conf, where I can set attribute

nat=yes

and this will ignore the IP and Port in the SIP headers and use the one for the SIP request and also waits for an incoming RTP stream to reply to.

I'd like to make use of this feature as we already have an Asterix server installed for AIX requests.

What would be the minimum configuration required for Asterix to act as the man in the middle on a new port as 5060 will still be used to connect directly to the SIP client? I don't care about authentication etc. I just need the Asterix to act as a SIP relay.

Thanks K

1
sipasterisk
Do you mean minimum hardware configuration? Any half decent server is good enough to run Asterisk with low call volumes. This isn't really a programming related question... – sipsorcery
Sorry it's not programming related. But it may be if I have to update the sip clients to overcome this issue. – kasuku

1 Answers

3
votes

You may also need to have "canreinvite=no" in the [general] section of your sip.conf. That setting keeps Asterisk in the call path, otherwise voice traffic may be sent directly from one endpoint to the other.