Re: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform

2005-03-20 Thread Kevin P. Fleming
Kevin P. Fleming wrote: If set to 'yes', and '183 Session Progress' has not already been sent, then '180 Ringing' is sent _and_ audio ringback is also generated (although I can't seem to figure out how that could work, since if '183 Session Progress' has not been sent, there is no early media in

Re: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform

2005-03-20 Thread Kevin P. Fleming
Tom Samplonius wrote: 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

Re: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform

2005-03-20 Thread Tom Samplonius
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. Calle

Re: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform

2005-03-19 Thread Kevin P. Fleming
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'

[Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform

2005-03-19 Thread Adam Rothschild
Hello, I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in turn is talking to an Asterisk server via SIP for call origination and termination. Seems simple enough, and it works for the most part, but: 1) Caller ID name data comes in on the PRI, but doesn't appear to get hande