Note, that was supposed to say: the only reason I can't do tone detect... ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, June 02, 2008 2:59 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] OZ question: disconnected number handling Anthony, Of course, you are correct. They are retarded, and they are sending progress inband. The only reason I can do tone detect or anything like it is because the originate app doesn't finish until the channel goes from PROGRESS to UP. About the only thing I can think of is to have OZ take an ALERT message with progress IE containing the "I'm a retard, here's inband messaging" info and have that move the channel state to UP. Or if there's another way for the originate app (or bridge app, depending on whether I'm using API or not) not to get snagged on this particular scenario then I'm all ears. I'll keep looking for other goofy cases like this. The application that I'm looking at right now is essentially a "disconnected number verifier" program. I'm trying to write an auto dialer that gets fed a list of phone numbers that we're pretty sure are disconnected and does its best to verify that fact. Of course, when the friggin' telco sends inband progress instead of a q931 message indicating an issue then we've got to play these silly games. Let me know what you think about the ALERT + PROGRESS inband = OZ channel up. -MC P.S. - when dialing out via a VoIP carrier, what kinds of things happen when a disconnected number is called? Does the VoIP carrier also have to deal with the retardation from the telcos? If so, how well does it work? Just curious if a SIP account would be good for this kind of application... ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, June 02, 2008 2:00 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] OZ question: disconnected number handling they are sending inband progress like retards. they are sending progress with media then playing sit instead of appropriate ISDN message. on trick you can do is try adding tone_detect app like some of the fax examples but for one of the 3 sit frequencies and transfer that call to hangup. On Mon, Jun 2, 2008 at 3:44 PM, Michael Collins <mcollins@fcnetwork.com> wrote: Guys, I don't know if this is "normal" or not so I'm hoping you can help me figure out what's up. I've got a disconnected number that I call on a PRI and the sequence goes like this: Dial number, hear ring back, see PROGRESS from telco, hear SIT and "you've reached a DC'd number...", then hear fast busy for 30 seconds (or so), then receive from telco that the call state is up, and then 3 seconds later call is terminated. I've PB'd the complete log here: http://pastebin.freeswitch.org/4549 FYI, I tried it with both with and without ignore_early_media. When ignoring early media I just get 60 seconds of FS internal ringback tone then I hear that the call failed. When not ignoring early media is when I get the SIT tone, etc. and that's also what is in pb 4549. Any thoughts on how to handle a call like this? FS doesn't consider the call "connected" because telco sends us PROGRESS_MEDIA but doesn't actually say that the call is "up" - at least as far as I can tell. Thanks, Michael _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400