[asterisk-users] asterisk server - no sound

[andre castro]

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