Hi all, I have decided to send this e-mail because you are the developer of Asterisk . We are developing a phone system using Asterisk as the VOIP gateway with 1 t410 PRI card and Sip Express Router as the proxy server but we have a problem. Our phone system setup like this: SIP phones------------>SER--------------->Asterisk---------------->PSTN(PRI connected to NORTEL DES 100 switch) transfer the call to Sip Express router then to the phone. So when there is a call from the pstn through asterisk and the phone is busy or the number is invalid ,asterisk tells the switch that the call is going on and the phone is ringing while it is not the case. I would like to find a way of informing Asterisk that the call is progressing or something like that, not ringing until it gets the correct message from SER . I am using Asterisk CVS-03/22/04-15:45:54 on Red Hat 9 and Sip Express Router version 12 on Red Hat 9. I tried to use PRI_causes but still the problem is there. Any idea on how I can solve this problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040609/e67e60e3/attachment.htm