On Thu, 2007-02-01 at 08:47 -0500, Matthew Rubenstein wrote: > The point is that the SIP carrier side gets the abort *before the SIP > carrier can complete the connection*. That doesn't take 45s. It takes > something like a few seconds. What is causing my (Asterisk) side to > abort right after completing registration? > > > On Thu, 2007-02-01 at 02:28 -0500, Asterisk wrote: > > Yeah, your waittime parameter in your call file is set to 45 seconds. > > > > db > > > > On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote: > > > I used the "FreePBX on Debian" HowTo at > > > http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles > > > to initiate calls to my SIP carrier. They get my registration, but they > > > see that my call is interrupted before they can complete the connection. > > > My Asterisk log shows that the call times out after the time (45s) > > > specified in my dialplan Dial() command. What is wrong? > > > > > > [from /var/log/asterisk/full]: > [...]
Alright, take a look the **Lines: **Line 1: Your dial sequence clearly shows the 45sec timeout value being applied as the second value in the dial plan "SIP/[EMAIL PROTECTED]|45| <<-- Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/[EMAIL PROTECTED]|45| M(say-call-2-digits^17182335097)g") in new stack **Line 2: The timer has expired 45000ms is the same 45 second timer that was set for timeout Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Nobody picked up in 45000 ms Line 3: The call is dropped towards the carrier. Maybe I am missing something here but it seems you are using a macro with some global variable set for a 45 second wait time for outbound calls. Thanks, Dave _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users