Hello Guys,

I have audio cut off at the begin of the verto call to  FreeSwitch that redirect to sip external voicemail (Access voicemail mailbox) . 

This happen when I use PCMU at verto codecs and sip codecs (if i use opus at verto codecs, there is no issue, but this causes audio transcoding) .

At dialplan i used the example "Bridging from WebRTC (mod_verto) to PSTN/ITSPs" from https://freeswitch.org/confluence/display/FREESWITCH/mod_verto.

I notice if i remove the playback action, there is no issue. But I need the playback action to send rtp packets to verto client.

I simulate this using another FreeSwitch as external voicemail server and I only listen "id followed by pound" from the initial message of voicemail ("Please enter your id followed by pound").

The log of this call is at https://pastebin.freeswitch.org/view/507fa115

What I can do to use PCMU at verto codecs and sip codecs on type of call?

Should i open a issue on FreeSwitch JIRA ?