Why asterisk drop call from H.323 to SIP


#1

I compiled a chan_oh323 0.6.6 and make it work with asterisk 1.0. I want to use asterisk as a gateway between H.323 and SIP. However, when I call from a H.323 phone to a SIP phone, the SIP phone rang correctly, but when I take up the SIP phone, the call is dropped after 1 second. Does anybody know the reason? thanks!!


#2

start * with a lot of v - maximum verbosity.
Look at it.
Place here most important part, and may be somebody will have some ideas.


#3

Hello,

I had the same problem but with a voip interconection bettwen Alcatel and cisco. The call was droped after 5 seconds. The problem was in that case a “Round trip delay” request from Alcatel, which Cisco does ont replay to. But you can be sure only after a verbose output in Asterisk.

Regards,
Mircea.