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 [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 > > [EMAIL PROTECTED] > > Bureau of Economic and Business Research > > University of Florida > > (352) 392-0171 Ext. 221 > > > > > > > > > >> -----Original Message----- > >> From: [EMAIL PROTECTED] > >> [mailto:[EMAIL PROTECTED] 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 > >>> [EMAIL PROTECTED] > >>> Bureau of Economic and Business Research > >>> University of Florida > >>> (352) 392-0171 Ext. 221 > >>> > >>> > >>> > >>> > >>> > >>>> -----Original Message----- > >>>> From: [EMAIL PROTECTED] > >>>> [mailto:[EMAIL PROTECTED] 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 > >>>> > >>>> _______________________________________________ > >>>> -- Bandwidth and Colocation Provided by > >>>> > >> http://www.api-digital.com -- > >> > >>>> 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 -- > >> > >>> 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 -- > >> > >> 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 -- > > > > 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
