Kevin, Thanks for the info. Clarification. The asterisk server is NOT on the same LAN as the phones. The asterisk server is in a datacenter only accessible via WAN.
However, all of the phones are in side of the same LAN. Will directmedia still function that way? Thanks David From: Kevin Larsen <kevin.lar...@pioneerballoon.com<mailto:kevin.lar...@pioneerballoon.com>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>> Date: Thursday, April 25, 2013 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>> Subject: Re: [asterisk-users] Sip and the media path You will want to look at the directmedia option. You will want all the phones on the same lan as the Asterisk server to be directmedia=yes and the ones on the wan to be directmedia=no. Then, internal calls will send the media between themselves without involving Asterisk, but ones outside on the wan will be forced to talk directly to the Asterisk server for everything. You might also want to look at the nonat option of directmedia. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 From: David Wessell <da...@ringfree.biz<mailto:da...@ringfree.biz>> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>>, Date: 04/25/2013 07:33 AM Subject: [asterisk-users] Sip and the media path Sent by: asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com> ________________________________ We're running asterisk 1.8 in the DC on a public IP address. Connecting to it are about 200 phones behind a LAN in a remote location. Is there a way to reliably keep asterisk out of the media stream on internal calls inside that LAN? All phones are Polycom Soundpoint phones. Asterisk would say in the media stream for any calls that traverse from LAN to WAN. However it would step out for LAN to LAN calls. Thanks David -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com<http://www.api-digital.com/> -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users