One way audio when using originate…

Home » Asterisk Users » One way audio when using originate…
Asterisk Users 2 Comments

We are having a problem when trying to use originate or AMI to make a
call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to
call the PSTN. When dialing from IP phones everything works fine. When
you try making the call with originate, AMI or a call file then the
remote person can hear you but you cannot hear them. Why would it
behave differently when dialing from a phone?

The server is behind NAT and uses externaddr to set the external IP
(static). Anyone had any experience with this?

Here is my (edited) sip.conf entry:

[libre-8793]
defaultuser=123456789
secret=XXXXXXXXX
fromuser=123456789
trustrpid=yes
sendrpid=yes
type=peer
fromdomain=i2next.com.mx
host=i2next.com.mx
nat=yes
qualify=no
insecure=port,invite
directmedia=no
disallow=all
allow=g729

2 thoughts on - One way audio when using originate…

  • Dear
    in normal mode, .call files make a call between the system and who you named
    remote person, I don’t know where are you?
    in natmode=yes, set qualify=yes.
    check the negotiated codecs also.
    Best

  • I know why this is happening. I had similar issues with an Avaya kit using SIP a while back. Took me a lot of investigation work to find my. Take a look at my One Way Audio article to see the fix. Let me know if this fixes the issue for you.