[Freeswitch-users] OZ question: disconnected number handling

Michael Collins mcollins at fcnetwork.com
Mon Jun 2 15:10:40 PDT 2008


Note, that was supposed to say: the only reason I can't do tone
detect...

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Monday, June 02, 2008 2:59 PM
To: freeswitch-users at 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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Monday, June 02, 2008 2:00 PM
To: freeswitch-users at 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 at 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 at 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 at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080602/985a1f88/attachment-0002.html 


More information about the FreeSWITCH-users mailing list