Hello,
I’m starting to use Asterisk.
Is it possible to put all outgoing calls on anonymous on the asterisk configuration, how can we do that (On which conf file and what is the parameter to edit) ?
Thank you in advance,
You can use https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CALLERID function to set the caller id information on those calls
But your provider may or may not honour that information. That will depend on the provider, the and the signalling protocol.
I tried to pass the callerid to “prohib” but the number is still displayed, below my extensions.conf file
[globals]
RTPFW1=SIP/RTPFW1
RTPFW2=SIP/RTPFW2
[XXX]
;Route to OUTSIDE
exten => _X.,1,NoOp(“XXX context”)
exten => _X.,n,NoOp(Variables: {CALLER})
exten => _X.,n,SIPAddHeader(Privacy: id)
exten => _X.,n,Set(CALLERID(num)=prohib)
exten => _X.,n,SIPAddHeader(Remote-Party-ID: "XXX" <sip:{CALLER}@XX.XX.XX.XX>;privacy=off;screen=no)
exten => _X.,n,SIPAddHeader(P-Asserted-Identity: tel:${CALLER})
exten => _X.,n,Wait(1)
exten => _X.,n,AGI(googletts.agi,“XXX”,fr)
exten => _X.,n,Dial({RTPFW1}/{EXTEN})
exten => _X.,n,Dial({RTPFW2}/{EXTEN})
Thanks,
Please markup your dialplan as pre-formatted text.
You should not manually add headers that will be automatically added by Asterisk.
You have not set the CALLERID function!
Does your provider allow you to override the caller ID? Do they accept RPID or PAI, and, if so, which?
Whilst noting that chan_sip is no longer effectively supports, and you should be using chan_pjsip, you should be setting sendrpid=yes or sendrpid=pai, according to which header your provider supports, and the appropriate num-pres and name-pres attributes on CALLERID.
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