[asterisk-users] TCP dial via proxy
Hello, We have an Asterisk dial which sends the call via a proxy using //, for example: Dial(SIP/${EXTEN}@peer_address//proxy_address) Does anyone know how we can make the SIP to the proxy use TCP? We tried making proxy_address match a peer in sip.conf with "transport = tcp" but that didn't seem to work. We are using chan_sip. Thanks very much for any advice. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Configuring Opus Forward Error Correction in Asterisk 16 (FreePBX)?
Hi. I am using Asterisk 16.27.0 in FreePBX 15.0.23.11. I installed via the FreePBX ISO (SNG7-PBX-64bit-2104.iso). I used the GUI to enable the Opus Codec in Asterisk SIP Settings and I can confirm the calls are using Opus 16000. How can I turn on fec. I am wondering if there might be a configuration file I need to modify to add “fec=yes” or something like that to turn this on. Thanks in advance for any help Brant Merryman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users