Thanks Danny :) I dont see there is any wrong with users.conf This is the contain of users.conf : ; ; User configuration ; ; Creating entries in users.conf is a "shorthand" for creating individual ; entries in each configuration file. Using users.conf is not intended to ; provide you with as much flexibility as using the separate configuration ; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the ; simple task of adding users. Note that creating individual items (e.g. ; custom SIP peers, IAX friends, etc.) will allow you to override specific ; parameters within this file. Parameter names here are the same as they ; appear in the other configuration files. There is no way to change the ; value of a parameter here for just one subsystem. ;
[general] ; ; Full name of a user ; fullname = New User ; ; Starting point of allocation of extensions ; userbase = 6000 ; ; Create voicemail mailbox and use use macro-stdexten ; hasvoicemail = yes ; ; Set voicemail mailbox 6000 password to 1234 ; vmsecret = 1234 ; ; Create SIP Peer ; hassip = yes ; ; Create IAX friend ; hasiax = yes ; ; Create H.323 friend ; ;hash323 = yes ; ; Create manager entry ; hasmanager = no ; ; Set permissions for manager entry (see manager.conf.sample for documentation) ; (defaults to *all* permissions) ;managerread = system,call,log,verbose,command,agent,user,config ;managerwrite = system,call,log,verbose,command,agent,user,config ; ; ; MAC Address for res_phoneprov ; ;macaddress = 112233445566 ; ; Auto provision the phone with res_phoneprov ; ;autoprov = yes ; ; Line Keys for hardphone ; ;LINEKEYS = 1 ; ; Line number for hardphone ; ;linenumber = 1 ; ; Local Caller ID number used with res_phoneprov and Asterisk GUI ; ;cid_number = 6000 ; ; Remaining options are not specific to users.conf entries but are general. ; callwaiting = yes threewaycalling = yes callwaitingcallerid = yes transfer = yes canpark = yes cancallforward = yes callreturn = yes callgroup = 1 pickupgroup = 1 ;nat = no ;[6000] ;fullname = Joe User ;email = j...@foo.bar ;secret = 1234 ;dahdichan = 1 ;hasvoicemail = yes ;vmsecret = 1234 ;hassip = yes ;hasiax = no ;hash323 = no ;hasmanager = no ;callwaiting = no ;context = international ; ; Some administrators choose alphanumeric extensions, but still want their ; users to be reachable by traditional numeric extensions, specified by the ; alternateexts entry. ; ;alternateexts = 7057,3249 ;macaddress = 112233445566 ;autoprov = yes ;LINEKEYS = 1 ;linenumber = 1 ;cid_number = 6000 Please advice and thanks :) On Tue, Apr 24, 2012 at 2:36 AM, Danny Nicholas <da...@debsinc.com> wrote: > Don't know about 1.8 but in 1.4 dahdi_genconf would update users.conf which > could mess with caller ID. Check your users.conf. If that's the problem, > you can fix and just do sip reload. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mc GRATH > Ricardo > Sent: Monday, April 23, 2012 2:31 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] HELP!! Caller ID "unknown" for all inbound call > (Satria Anamarta) > > Hi > > I'm not agree problem could be cause from IRQ setting, I think in that way > problem should be more unstable, moreover no voice communication problem > with DAHDI service start up. > Best regards > > Mc GRATH Ricardo > E-Mail mcgra...@mail2web.com > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New > to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users