Audio cutting off after ring

Asterisk Asterisk SIP
1

Hi all,

I am trying to set up a basic oubound call from asterisk v16.
Basically, I hear the first ring, but the line goes silent after that, but the call is received, answered and can be hanged up succesfully.

I do not have NAT enabled (to my knowledge) and I don’t see any errors showing up in the logs regardign the audio.

I’m not sure if it is a codec issue (I allow alaw/ulaw/gsm) and the SIP message shows Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw), so I think the codec is fine as well.

Below is my sip.conf file

[general]
context=from-trunk
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
;tcpbindaddr=0.0.0.0
allowguest=no
transport=udp
srvlookup=yes
mohinterpret=default
mohsuggest=default
trustrpid = yes
sendrpid = yes
sendrpid = pai
canreinvite=no
directmedia=yes
language=en
allowguest=no
nat=no
disallow=all
port=5060
bindaddr=0.0.0.0
qualify=yes
disable=all
allow=alaw
allow=ulaw
allow=gsm
insecure=invite,port
dtmfmode=auto
srvlookup=yes

Any help will be appreciated, I can upload call loges if necessary, thanks

2

same => n,Set(CALLERID(name)=+${CALLERID(name)})
same => n,Wait(1)
same => n,Dial(SIP/OUTBOUND/99904${EXTEN},45,rT)

This is the dialplan that gets hit, maybe the issue is in the Dial line?

3

Asterisk 16 is no longer supported.

chan_sip is no longer supported and is not in the source code of the latest two versions of Asterisk.

When using chan_sip, having this in the general section is dangerous.

You haven’t provided any logs, but my guess is that you’ve received a 183 response, and either no media, or early media is blocked somewhere. You may need to explicitly call Progress(), to ensure that early media is passed, or you can use Dial options to force ringback, even if there is early media.

These represent conflicting settings for the same option. The use of canreinvite for this option was deprecated about a decade ago.

This is not a documented parameter.

This is an often misunderstood setting, and the default,. “auto”, is almost always suitable.

Your sip.conf file is incomplete. It doesn’t define any peers, let alone one called OUTBOUND.

4

rtpkeepalive = 1