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<p class="MsoNormal"><span lang="EN-US">Hello, I have problems with calls between asterisk and avaya with canreinvite=yes enabled and Direct IP-IP? y (on avaya).<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Direct rtp work well but when avaya side client press hold,transfer or conferens, avaya use deleyed offer (INVITE without SDP). Asterisk don’t resend this INVITE to client, just send 100 and 200 ok with SDP and get back
ACK with new session SDP from avaya side (it use medpro board for music on hold). Asterisk don’t send this information to client softphone because invite from avaya without SDP. Avaya send rtp from medpro to asterisk softphone and asterisk softphone still
send rtp to avaya side phone directly.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">After avaya phone release hold button, avaya send again INVITE w/o SDP, asterisk send 100 and 200 ok to avaya, and then send INVITE to asterisk’s softphone with SDP, but this SDP it gets from previous ACK. Actual Information
from avaya backed with ACK again, but asterisk don’t tell about it to his softphone.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">I hear that cisco system have some special settings to fix it by enabling early offer. Avaya – don’t.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Is it possible force asterisk to use delayed offer?<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US">Sorry for my English.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-US"><o:p> </o:p></span></p>
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