[asterisk-users] asterisk server - no sound

Marcelo Terres mhterres at gmail.com
Tue Jun 6 12:31:34 CDT 2017


Try to use the echo app. If you can listen your echo, so it is
something in the network.

Regards,
Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On 6 June 2017 at 14:18, andre castro <andre at andrecastro.info> wrote:
> 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
>
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