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<font size="-1"><font face="Arial">hi,<br>
<br>
i have the same issue, i think its related to slow audio setup
during the call<br>
<br>
<br>
</font></font>
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</style>Regards,<br>
Bipin<br>
<br>
<br>
<hr>
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<div class="moz-cite-prefix">-------- Original Message --------<br>
Subject: [Freeswitch-users] Audio cut off at the begin of the
verto call to sip external voicemail<br>
From: José Lopes <a class="moz-txt-link-rfc2396E" href="mailto:jose.lopes@itcenter.com.pt"><jose.lopes@itcenter.com.pt></a><br>
To: FreeSWITCH Users Help
<a class="moz-txt-link-rfc2396E" href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a><br>
Date: 1/5/2017, 6:35:45 PM<br>
</div>
<blockquote
cite="mid:CAK5rbQtOK9ZXm+uYc+VZaZRE71M=Qg5cknj1WLevCQxa09goCw@mail.gmail.com"
type="cite">
<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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 face="verdana, sans-serif" size="2">Best
regards,</font></div>
<div><font face="verdana, sans-serif" size="2">Jose
Lopes<br>
</font></div>
</div>
</div>
</div>
</div>
</div>
<br>
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<pre wrap="">_________________________________________________________________________
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</blockquote>
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