<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.2800.1400" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Hi
everyone,</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Does anyone know the
answer for this situation? I have Asterisk with E1 PRI links, with SIP phones
registered to Asterisk and with h.323 connection to Cisco CallManager. I am
using oh323. I think I have a problem with codecs but I do not know exactly what
is wrong.</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>this is working
ok:</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2>--------------------------</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Call from
CallManager (7960) to SIP phone on Asterisk (X-Lite or 7960 with SIP image)
- working OK</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Call from
CallManager (7960) to E1 PRI trunk to PSTN through Asterisk - working
OK</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Call from E1 PRI
trunk from PSTN through Asterisk to CallManager (7960) - working
OK</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>here is the
problem</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2>---------------------------</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Call from SIP phone
to CallManager - rings the phone, in the moment when called party picks the
receiver Asterisk crashes with core dump</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>Interesting is that
if you establish a call in opposite direction (from CallManager to SIP phone)
prior to that one, Asterisk wouldn't crash sometimes</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial size=2>I will appreciate if
anyone can help</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2>Tomica</FONT></SPAN></DIV>
<DIV><SPAN class=286473422-14022004><FONT face=Arial
size=2></FONT></SPAN> </DIV></BODY></HTML>