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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I am using asterisk-1.4.21 and it is configured to pass
media through it for SIP calls. I have observed that if the callee answers the
call and starts speaking immediately for e.g. ‘Hello one two three’,
the caller would get to hear only ‘one two three’. From packet
captures I can see that asterisk receives all the RTPs from the callee but it truncates
the ‘Hello’ word from the voice path when passing the stream on the
other side.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The signaling gets complete between caller and callee, so
asterisk should bridge the channels immediately. I am using canreinvite=no and
nat=yes option in sip.conf.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Has anyone observed this issue why asterisk is cutting of
the initial voice? <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><em><i><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial;font-style:normal'>---</span></font></i></em><em><i><font
size=2 color="#000066" face=Arial><span style='font-size:11.0pt;font-family:
Arial;color:#000066;font-style:normal'>Mayur</span></font></i></em><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
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