We have a customer using ConfBridges. Party A is connected, audio is fine. We originate a call to party B through an Avaya switch. It forwards the call to IVR. The two channels are added to the same ConfBridge. Using a wireshark capture, I can listen to the audio for both channels. Initially, everything on the audio sounds great. Audio for the connection to the Avaya switch always sounds fine. Audio that party A hears becomes garbled about 7-8 seconds into the prompts playing from the IVR. I can definitely hear the audio from voice mail plays one prompt when it's fine. Then, it changes to a different voice/prompt that is louder. At that point, the audio party A hears is garbled. This is using chan_sip for both channels. We have no problems with audio calls to other numbers, only to the IVR Only reason we are using chan_sip is because we need to use REFERs and are working on a patch submission (hoping asterisk 16.5.0) to resolve a PJSIP REFER issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190605/e8051d15/attachment.html>