On Sat, 19 Mar 2005 17:08:59 -0700, Kevin P. Fleming <[EMAIL PROTECTED]> wrote: > Adam Rothschild wrote: > > > 1) Caller ID name data comes in on the PRI, but doesn't appear to get > > handed off to the Asterisk server via SIP, at least not in any > > format that Asterisk understands. Caller ID _number_ works fine. > > > > (I'm guessing this has something to do with the 'remote-party-id' > > header, but I've tried with it both enabled and disabled in the > > 'sip-ua' IOS configuration stanza.) > > Turn on RPID generation in your AS5300, and then set "trustrpid=yes" in > the SIP user entry for the gateway in Asterisk. > > > 2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk > > -> Dial([...],,r) calls placed. Music on hold ([...],,m) works > > fine. > > Play with the 'progressinband' setting in Asterisk to see if you can > affect this; it has three settings 'yes', 'no', and 'never'. It's likely > that what is happening is that Asterisk is sending '180 Ringing' and the > AS5300 is not generating ringback itself or asking the switch on the > other end of the PRI to do it (or farther up the PSTN chain).
What does "progressinband" do exactly? Does it disable 180 responses? I can't find any references to what effect "no", "yes", and "never" have on the SIP exhange. In fact, why is it called "inband" if it involves the SIP messages? Wouldn't "inband" refer to messaging in the media stream (RTP)? Tom _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
