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 ? Best regards, Jose Lopes