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