Incomming calls are on TDM lines connected to the Digium card. Calls between extentions are on the LAN for SIP registered users/ip phones.
Gary Baribault On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote: > > Incoming and outgoing calls are on SIP or on ZAP? > > Zeeshan A Zakaria > > -- > Sent from my Android phone with K-9 Mail. > >> On 2010-06-01 3:28 PM, "Gary Baribault" <g...@baribault.net >> <mailto:g...@baribault.net>> wrote: >> >> 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