My assumption is that inbound/outbound calls are DAHDI and that internal calls are SIP. Can OP post "core show channels" from working and non-working calls?
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Baribault Sent: Tuesday, June 01, 2010 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] no sound between extensions Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s. The 6001 rings, but when the phone is picked up, I have no sound. I have the same problem between any phones in the system, but this is the simplest example. Incoming calls and outgoing calls work fine, sound is correct. Voice mail works fine as well, the IVR works great. Any ideas? Gary Baribault -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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