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<DIV> Hello list, I have been working with asterisk for a couple of months and now I have run into a problem, I have the following setup</DIV>
<DIV> </DIV>
<DIV>PSTN <======>Asterisk(remote behind nat)<=======>IAX<======>Asterisk(local public ip)<========>OH323<====>Gateway</DIV>
<DIV> </DIV>
<DIV>I want to terminate incoming calls from the gateway in the remote asterisk.</DIV>
<DIV> </DIV>
<DIV>My problem is that when I started testing this setup the calls coming to the pstn lines from sip and h323 clients worked, now that I want to terminate my calls in the pstn lines coming from the gateway the calls just hangup with no error message and just a message saying no one is available to answer.</DIV>
<DIV> </DIV>
<DIV>I have been using my own phone lines in the office for testing, and this happens every time that the call is passed and the phone where the call is supposed to land receives the call, in that moment asterisk hangs for no apparent reason, the asterisk with the zap channels is stable v.1.0.3 and the one with OH323 is cvs with openh323 and pwlib Janus patch and OpenH323 v.0.7.0.</DIV>
<DIV> </DIV>
<DIV>If anyone has done something similar I would appreciate the help, any clues or extra information you need I would gladly send.</DIV>
<DIV> </DIV>
<DIV>Dan Flores</DIV></TD></TR>
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