hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints:
Peer User/ANR Call ID Format Hold Last Message Expiry Peer peer.ip 1001 1...-5060 (ulaw) No Rx: ACK 1001 But I hear nothing at the peer's end. When one peer calls another, sound comes through just fine. So my hunch is that is something to do with the audio supplied by the server. Do I need to have alsa installed?? Any hint? sip.conf: [general] context = unauthenticated bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes videosupport = no textsupport=yes alwaysauthreject=yes allowguest=no [1001] ; grandstream 1 context = home type = friend callerid = One <1001> secret = XYZ host = dynamic mailbox = 1001 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport [1005] ; mobile context = home type = friend callerid = Five <1005> secret = XYZ host = dynamic mailbox = 1005 disallow = all allow = ulaw transport = udp dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically nat=force_rport extensions.conf: [home] exten = 102,1,Answer() same = n,Wait(1) same = n,Playback(beep) same = n,Wait(1) same = n,Playback(im-sorry) same = n,Wait(1) same = n,Playback(number-not-answering) same = n,Wait(1) same = n,Hangup() exten => 1001,1,Dial(SIP/1001) ; grandstream phone exten => 1005,1,Dial(SIP/1005) ; mobile -- oooooooooo.io bibliotecha.info -- _____________________________________________________________________ -- 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