We're setting up a SipXecs server in-house to manage about 20-30 polycom sip phones. We have an Audiocodes Mediant 2000 to use as a gateway but for testing I was also trying to setup sip in/out dialing through the firewall. I've wanted a reason to start playing with freeswitch so I thought this would be an excellent opportunity to use freeswitch for the Nat traversal.
I've been through the wiki and reviewed list archives but I'm missing something. I have RC3 on Centos (initially a trixswitch load but then upgraded to the new RC3) with the standard config files. I did remove the older ones and re-installed the samples. This is a pretty basic install with a gafachi gateway setup for the outbound sip profile, and the firewall's external ip setup for the external_rtp and external_sip values (in vars.xml), and the firewall port forwards all recommended ports(from wiki getting started page) into freeswitch. This is where I'm stuck. I have sipx attempting to send calls to Freeswitch on port 5070 (for nat) but Freeswitch won't accept the call and is logging: 2008-04-24 16:20:26 [DEBUG] sofia.c:219 sofia_event_callback() event [nua_i_state] status [407][Proxy Authentication Required] session: n/a The nat sip_profile is setup per default to answer port 5070 and authentication (per default) is disabled. I'm sure it's something obvious but what am I missing? Thanks, Jay
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