<div dir="ltr"><div>Hello Guys,</div><div><br></div><div>I have audio cut off at the begin of the verto call to FreeSwitch that redirect to sip external voicemail (Access voicemail mailbox) . </div><div><br></div><div>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) .</div><div><br></div><div>At dialplan i used the example "Bridging from WebRTC (mod_verto) to PSTN/ITSPs" from <a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_verto">https://freeswitch.org/confluence/display/FREESWITCH/mod_verto</a>.</div><div>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.</div><div><br></div><div>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").</div><div>The log of this call is at <a href="https://pastebin.freeswitch.org/view/507fa115">https://pastebin.freeswitch.org/view/507fa115</a></div><div><br></div><div>What I can do to use PCMU at verto codecs and sip codecs on type of call?</div><div>Should i open a issue on FreeSwitch JIRA ?</div><div><br></div><br clear="all"><div><div class="gmail_signature"><div dir="ltr"><div dir="ltr"><div dir="ltr"><font size="2" face="verdana, sans-serif">Best regards,</font></div><div><font size="2" face="verdana, sans-serif">Jose Lopes<br></font></div></div></div></div></div>
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