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<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>I posted a
problem earlier thinking it was due to a lack of sound card. Several
members stated that you do not need a sound card to play audio to a PRI
channel. I did some further testing and discovered that there is a problem
with call progress tones or signaling on my PRI. I think that
the reason I am not hearing audio from the MeetMe() or Playback() apps. is
because the the calling side of the PRI (NEC IPX), is not seeing the Answer
signal. I believe it is waiting for a ring and/or
answer condition even after Asterisk has executed an Answer() and
Playback().
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>The only other
problem that I am having with my setup is that the CONSOLE/DSP is not
functional... I am not sure if the two problems are related. Any help is
appreciated. Please see my two examples
below:</SPAN></FONT></DIV></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004></SPAN></FONT><FONT
face=Arial size=2><SPAN class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>Unless my incoming
DID (2000), is pointed to a SIP station that is registered and
functional, I do not receive call progress tones on inbound
calls.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>If I point the DID
to an application like:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004> [inbound_pri]<BR>; PRI from the
NEAX2400</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>exten =>
2000,1,Wait,3<BR>exten => 2000,2,Answer<BR>exten =>
2000,3,MeetMe,|Mps<BR>exten => 2000,4,Hangup</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>I will not hear any
initial ringback, and once answered there will be no audio on the
channel.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>If I point the DID
to a registered SIP station like:</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>[inbound_pri]<BR>;
PRI from the NEAX2400</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>exten =>
2000,1,Wait,3<BR>exten => 2000,2,Dial,SIP/2001,15,Tr<BR>exten =>
2000,Hangup</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004>It will provide
ringback tone to the calling channel on the PRI, and when the ringing
SIP phone answers there will be 2-way speech path.</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
class=546301419-29082004></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN class=546301419-29082004> </DIV>
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