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There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

I have a SIP phone that can make calls but can't recieve calls. Can anyone suggest why? sip show peers: Name/username Host Dyn Nat ACL Mask Port Status 601/601 (Unspecified) D N 255.255.255.255 0 UNKNOWN -- Executing Dial("SIP/206.132.91.139-0814c9f8", "SIP/601|20|r") in new stack Jul 18 15:50:18

I have a SIP phone that doesn't support three-way calling. Is there a way to do three-way calling from a SIP phone server side instead? TKS __________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250

I have a SIP phone that is registered. i can make calls out from the phone. I can't make calls to the phone. What does the error message mean? How can I fix it? Thanks! 8 headers, 0 lines Destroying call '6b9fb03c4677b9266e1263fb0c7ea304@127.0.0.1' Jul 15 22:10:49 NOTICE[262159]: rtp.c:285 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible == CDR updated

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten