Thanks for the tip about sip set debug <peer>. I was able to capture some information about the call in progress, but I am confused as to what I see. When I pick up my sip phone I dial *811<area_code><prefix><number>, and the first invite I see is going to 1<area_code><prefix><number>@<my_asterisk_ip>. Shouldn't the *81 be included in the request?
Is it possible that the linksys pap2 that I am using is removing the *81 prior to placing the invite request? John John Millican wrote: > John Koenig wrote: > >> I tried all of the suggestions, and still the callerid remains intact. >> I guess at this point I am starting to wonder what bit of logic is being >> run when I dial *8111XXXXXXXXXX... >> >> Is there a way I can trace how a call is being processed within >> asterisk? Or even see what I am sending to my VoIP terminating node? >> >> John >> >> John Millican wrote: >> >>> Doug Lytle wrote: >>> >>> >>>> John Koenig wrote: >>>> >>>> >>>>> exten=s,1,set(CALLERID(all)= null) >>>>> exten=s,n,Dial(${ARG1}) >>>>> >>>>> >>>>> >>>> Just a guess. >>>> >>>> exten => s,1,Set(CALLERID(all)= null <0>) >>>> exten => s,n,SetCallerPres(prohib) >>>> exten => s,n,Dial(${ARG1}) >>>> >>>> >>>> Doug >>>> >>>> >>>> >>> I believe you need to use: >>> exten => s,1,Set(CALLERID(all)=) >>> To set an empty callerId >>> >>> >>> > > typing: > sip set debug peer <peer_name> > at the CLI will give you a bunch of information as to what is going on > with that peer > > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users