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i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

I seem to be encountering a problem with the db.h file when compiling gastman. Incorrect definitions for functions and so on. Here are the errors: If I don't copy the db.h from asterisk source to the include directory of gastman I get: gui.c:31:16: db.h: No such file or directory If I do then I get: gui.c:743: too many arguments to function This is for the line: if (!(res =

Anyone know how to specify a call group (as specified in the example zapata.conf) when using Dial? No other reference I can find. Jim Friedeck

Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/