Good to hear you have found a temporary solution, although I think it is the permanent solution.
Keypad protocol is a bandaid to fix the real problem, not a solution. The problem is that many PBXs cant set the caller ID presentation fields, so keypad protocol was derived to work around it and allow end users to modify caller ID representation from the handset 'keypads' Use the caller ID presentation flags! That is the RIGHT way to do things on ISDN. Damon > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Michel Koenen > Sent: Monday, August 29, 2005 3:45 PM > To: [email protected] > Subject: [Asterisk-Users] How to use * and # as part of number > indialcommand > > > From: "Damon Estep" <[EMAIL PROTECTED]> > > Subject: RE: [Asterisk-Users] How to use * and # as part of number > > indialcommand > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <[email protected]> > > Message-ID: > > <[EMAIL PROTECTED] > systems.com> > > Content-Type: text/plain; charset="US-ASCII" > > > > Michel > > > > Send me the same output for a dial string that only sends the *31* > > > > Is this an ISDN line? What type of card/signalling/switchtype are you > > using? > > > >[snip] > > Hi Damon, > > Thank you for your extensive answer. The SetCallerPres(prohib) in > combination with adding 'usecallingpres=yes' into zapata.conf does the > job to set CLIR so I am helped out (for now). I've added this to the > voip-ip wiki at SetCallerPres. > > I see this as a kind of workaround because I am still trying to figure > out why the keypad protocol is not working as it should be. > > To come back on your questions: > Yes it is an ISDN line, it is an ISDN card with HFC chipset, I use > asterisk 1.0.7 bristuffed with zaphfc module. > Switchtype: euroisdn > Signalling: bri_cpe_ptmp (immediate=no, overlapdial=yes) > Internally I have also ISDN HFC but in NT mode, this is where i > originate the call from. In the logging this is the Zap/2-1 channel. > > Here is the log when I use the *31* in the dial command: > > -- Accepting voice call from '1001' to 's' on channel 0/2, span 1 > -- Executing NoOp("Zap/2-1", "") in new stack > -- Executing Answer("Zap/2-1", "") in new stack > -- Executing Playtones("Zap/2-1", "dial") in new stack > -- Executing ResponseTimeout("Zap/2-1", "15") in new stack > -- Set Response Timeout to 15 > == CDR updated on Zap/2-1 > -- Executing Dial("Zap/2-1", "Zap/4/*31*") in new stack > -- Requested transfer capability: 0x10 - 3K1AUDIO > -- Called 4/*31* > -- Zap/4-1 is making progress passing it to Zap/2-1 > -- Channel 0/1, span 2 got hangup > Aug 29 23:18:45 WARNING[6591]: app_dial.c:412 wait_for_answer: Unable > to forward voice > -- Hungup 'Zap/4-1' > == No one is available to answer at this time > > The debug log shows this (I wonder where the conferencing log is > coming from because I don't use conferencing ??) > > Aug 29 23:18:45 DEBUG[6591]: disabled echo cancellation on channel 4 > Aug 29 23:18:45 DEBUG[6591]: Set option TDD MODE, value: OFF(0) on Zap/4-1 > Aug 29 23:18:45 DEBUG[6591]: Updated conferencing on 4, with 0 conference > users > Aug 29 23:18:45 DEBUG[6591]: Set option AUDIO MODE, value: OFF(0) on > Zap/4-1 > Aug 29 23:18:45 DEBUG[6591]: disabled echo cancellation on channel 4 > Aug 29 23:18:45 VERBOSE[6591]: -- Hungup 'Zap/4-1' > Aug 29 23:18:45 VERBOSE[6591]: == No one is available to answer at this > time > Aug 29 23:18:45 DEBUG[6591]: Exiting with DIALSTATUS=NOANSWER. > > > > Best regards, > Michel > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
