PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same.
On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson <dicken...@cfmc.com> wrote: > Does Asterisk, currently using version 1.4, get any more information about > the result of an outbound call made over a PRI line compared to a call via a > SIP trunk? > > As an example, in a PRI call there is this message that shows up on the > console: > > [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. > > for a call to a fax machine. Does asterisk set anything that a dialplan can > access that can know the call was to a fax machine? > > If a call is placed to a number that is disconnected so a special information > tone is played can either a PRI call or a SIP call know this without > analyzing the audio stream? > > Are there reasons to prefer the use of PRI over SIP or SIP over PRI? > > I would like people's opinions as to if one form is better than the other in > any meaningful way. > > Thanks for you feed-back. > -- > Jim Dickenson > mailto:dicken...@cfmc.com > > CfMC > http://www.cfmc.com/ > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
