Garbled Voicemail on SIP calls from Asterisk Extension

We have an Asterisk server running Asterisk 13 LTS and the server utilises a SIP gateway service from an external telecommunications provider for all external calls. Internal calls along with external inbound and outbound external calls are operating fine except for an issue with one phone extension. For one particular extension the user can make and recieve voice-calls without an issue to external landlines and mobiles however whenever this user make an external call and leaves a voicemail on some phones the voicemail is garbled when the reciever plays the voicemail back.

The issue doesn’t occur everytime the user leaves a voicemail on receiptants phone.

I’ve run test calls from the users extension and got debug of a call when the voicemail is garbeled and another when the voicemail is not. The “set debug peer ###” seem to be the same with no specific difference or additional log messages.

Is there a particular way to debug the portion of SIP call when on an Asterisk server when a SIP session from the Asterisk Server is leaving a voicemail on an external system?

The SIP portion doesn’t carry audio, the RTP portion does. You’d need to do a packet capture or look at “rtp set debug on” to see if anything looks out of place. Asterisk generally acts as a packet forwarder though - it will alter the media if it is transcoding but otherwise passes things along as received.

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