One-way audio issue: Opus ↔ G.711 (u-law) in Asterisk

Asterisk Asterisk Support
1

Hi everyone,

I’m facing a one-way audio problem in my Asterisk setup when transcoding between Opus and G.711 u-law:

  • Opus user → Asterisk → u-law user works fine.

  • u-law user → Asterisk → Opus user results in one-way audio (Opus side hears nothing)

Environment

  • Asterisk version: (18.26.1)

  • Opus codec module: codec_opus.so (Running)

  • SIP channel driver: chan_sip

  • NAT: Yes

  • canreinvite=no (directmedia=no)

What I’ve Checked

  • Opus codec modules are loaded (codec_opus.so, format_ogg_opus.so, res_format_attr_opus.so).

  • sip set debug on shows both codecs in SDP.

  • NAT settings configured (externip, localnet).

  • canreinvite=no to force Asterisk in media path.

2

Is Asterisk from a package?

Does “rtp set debug on” show media flowing?

What endpoints are in use?

3

I have installed the Asterisk from source and media is flowing (Whatsapp—-asterisk—-Voipswitch)