[asterisk-users] ISDN Cause codes

Martin Smith martins at bebr.ufl.edu
Mon Nov 24 07:54:03 CST 2008


Hi Robert & all,

Maybe someone else can speak to using Progress(), but I don't know if it
is required or not. On our system, we didn't need it, and these settings
below (plus a call to the telco to tell them to turn on operator
messages, don't eat them) did the trick.

Good luck,

Martin Smith, Systems Developer
martins at bebr.ufl.edu
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Robert Boardman
> Sent: Saturday, November 22, 2008 11:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] ISDN Cause codes
> 
> I have found that the messages are not played as the hangup 
> cause clears 
> down the channel and passed hangup to the other end
> 
> should I have progress() before the dial command?
> 
> Robb
> 
> Martin Smith wrote:
> > Hi Robert,
> >
> > I'd recommend the following options for Dial() so that you 
> corroborate
> > operator messages w/ cause codes:
> >
> >  1. remove R and r - we've found this can supress operator 
> recordings on
> > early audio
> >  2. likewise, remove m to disable MOH
> >
> > Also, check the values of DIALSTATUS to compare to HANGUPCAUSE.
> >
> > Good luck,
> >
> > Martin Smith, Systems Developer
> > martins at bebr.ufl.edu
> > Bureau of Economic and Business Research
> > University of Florida
> > (352) 392-0171 Ext. 221 
> >
> >  
> >
> >   
> >> -----Original Message-----
> >> From: asterisk-users-bounces at lists.digium.com 
> >> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> >> Robert Boardman
> >> Sent: Friday, November 21, 2008 3:07 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [asterisk-users] ISDN Cause codes
> >>
> >> Thanks for the reply
> >>
> >> Could you be a little more specific?
> >>
> >> Thanks
> >> Robb
> >>
> >> Martin Smith wrote:
> >>     
> >>> Hi Robert,
> >>>
> >>> I'd suggest tweaking the Dial() arguments so that you (1) 
> >>>       
> >> allow early
> >>     
> >>> audio, (2) don't force it play ringing to the calling 
> party, and (3)
> >>> modify any other options to be as relaxed as possible. if 
> >>>       
> >> you make those
> >>     
> >>> changes, you'll start hearing the operator message 
> >>>       
> >> recordings and those
> >>     
> >>> are sometimes easier to reference against the cause codes.
> >>>
> >>> Cheers,
> >>>
> >>>
> >>> Martin Smith, Systems Developer
> >>> martins at bebr.ufl.edu
> >>> Bureau of Economic and Business Research
> >>> University of Florida
> >>> (352) 392-0171 Ext. 221 
> >>>
> >>>  
> >>>
> >>>   
> >>>       
> >>>> -----Original Message-----
> >>>> From: asterisk-users-bounces at lists.digium.com 
> >>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> >>>> Robert Boardman
> >>>> Sent: Thursday, November 20, 2008 5:56 PM
> >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>>> Subject: [asterisk-users] ISDN Cause codes
> >>>>
> >>>> Hi All
> >>>>
> >>>> Just been looking at stats for one of my sites, and I'm 
> >>>> conserned about 
> >>>> the number of error cause codes being returned from the telco
> >>>>
> >>>> for example
> >>>>
> >>>> 12000 calls processed
> >>>>
> >>>> 131 are cause code 31* normal. unspecified.*
> >>>>
> >>>> 139 are cause code 28 * invalid number format (address 
> >>>>         
> >> incomplete).*
> >>     
> >>>> 112 are cause code 1 *Unallocated (unassigned) number.
> >>>>
> >>>> *this adds up to about 3% of calls not completing.
> >>>>
> >>>> there are various other codes including 17 busy 34 channel 
> >>>> unavaliable 
> >>>> and 44 requested channel unavaliable, which add up to 
> another 1%.*
> >>>> *
> >>>> the telco says there is no problem with the line, I'm trying to 
> >>>> understand what the problem could be
> >>>>
> >>>> now  alot of calls complete OK so I don't think is my configs
> >>>>
> >>>> Any advice would be appriciated
> >>>>
> >>>> Versions
> >>>> asterisk 1.4.21.1
> >>>> zaptel 1.4.12.1
> >>>>
> >>>>
> >>>> Robb
> >>>>
> >>>> _______________________________________________
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