Correct. You don't supply a sbc if sipxbridge is oing that. So that's correct.
============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: Discussion list for users of sipXecs software <[email protected]> Sent: Tue Oct 12 11:53:38 2010 Subject: Re: [sipx-users] SIP URI Dialing >You will either need to use sipxbridge and Ahhhhh... But how do I do that? On the "Internet Calling" page, there is a drop down for "Default SBC". Sipxbridge is not in the list. The only choice is "none". >configure your firewall to allow the ports needed I do have sipxbridge working fine through our firewall with a couple of different ITSP's. Thanks for the reply, Scott. From: Tony Graziano [mailto:[email protected]] Sent: Tuesday, October 12, 2010 8:38 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] SIP URI Dialing Internet calling (uri dialing) is the same as trunking in that it needs a SBC in order to facilitate a transaction over NAT. You will either need to use sipxbridge and configure your firewall to allow the ports needed or use an SBC and configure sipx to allow internet calling via the external SBC. It sounds as though your firewall is either trying to proxy the media or is not configured for it at all if you are using sipxbridge. On Tue, Oct 12, 2010 at 8:33 AM, Scott Richesson <[email protected]> wrote: --- Repost: Sorry if you've already seen this message --- Hello, We are using sipx 4.2.1 and have sipxbridge working fine for a couple of outbound trunks. Our SIPX server and phones are behind a NAT. I have been playing around with SIP URI dialing, and I am not sure how the media relaying/NAT busting is supposed to work. In my experimenting, I can get a call to connect, but I get no RTP. Upon looking further, it looks as if the outgoing RTP is being directed directly from the phone to the remote URI with no relay. (I would rather the RTP be relayed so that I don't have to open up my firewall for the phones). I am receiving no incoming RTP at all, which suggests to me that there is no NAT compensation being done. Anyway, I don't have any immediate need for URI dialing, but am just curious if this is supposed to work, or if there is any way to configure the behavior here. Thank You, Scott Richesson IT Manager Cincinnati Fan _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
