[asterisk-bugs] [JIRA] (ASTERISK-28761) Assigning CallerIDNum from DNIDDigits to chan_pjsip

[Vyrva Igor (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28761?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249865#comment-249865 ] 

Vyrva Igor commented on ASTERISK-28761:
---------------------------------------

PJSIP:
{code}
[phone-endpoint](!)
 type = endpoint
  context = local
  disallow = all
  allow = ulaw,alaw
  dtmf_mode = rfc4733 
  direct_media = no
  rtp_symmetric = yes
  language = ru
  asymmetric_rtp_codec = yes

[phone-auth](!)
 type = auth
  auth_type = userpass
  password = 123456q

[phone-aor](!)
 type = aor
  max_contacts = 1 
  default_expiration=600
  qualify_frequency=150
  authenticate_qualify=yes

[422](phone-endpoint)
 auth = 422
  aors = 422
[422](phone-auth)
 username = 422
[422](phone-aor)

[423](phone-endpoint)
 auth = 423
  aors = 423
   send_rpid=yes
  device_state_busy_at=2 ; The number of in use channels which will cause busy to be returned as device state (default: «0»)
[423](phone-auth)
 username = 423
[423](phone-aor)
 max_contacts=2
{code}

> Assigning CallerIDNum from DNIDDigits  to chan_pjsip
> ----------------------------------------------------
>
>                 Key: ASTERISK-28761
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28761
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 16.8.0
>         Environment: CentOS 7
> Asterisk 16.8 (src)
>            Reporter: Vyrva Igor
>            Severity: Minor
>
> If the channel that responds to the Dial/Queue (dstchannel) call is running on chan_pjsip for it, the CallerIDNum parameter takes the value from the DNIDDigits parameter of the call Initiator (channel)
> Channel:
> {code}
> Name= SIP/423-00000031
> Type= SIP
> UniqueID= 1582885712.361
> LinkedID= 1582885712.361
> CallerIDNum= 423
> CallerIDName= AP-423
> ConnectedLineIDNum= (N/A)
> ConnectedLineIDName=(N/A)
> DNIDDigits= 300
> {code}
> Dstchannel:
> {code}
> Name= PJSIP/666-00000003
> Type= PJSIP
> UniqueID= 1582885712.364
> LinkedID= 1582885712.361
> CallerIDNum= 300
> CallerIDName= (N/A)
> ConnectedLineIDNum= 423
> ConnectedLineIDName= AP-423
> DNIDDigits= (N/A)
> {code}
> However, if the meeting works in chan_sip that happens
> Channel
> {code}
> Name= SIP/423-00000036 
> Type= SIP 
> UniqueID= 1582896505.420 
> LinkedID= 1582896505.420 
> CallerIDNum= 423 
> CallerIDName= AP-423 
> ConnectedLineIDNum= (N/A) 
> ConnectedLineIDName=(N/A) 
> DNIDDigits= 300
> {code}
> Dstchannel
> {code}
> Name= SIP/422-00000037 
> Type= SIP 
> UniqueID= 1582896505.423 
> LinkedID= 1582896505.420 
> CallerIDNum= (N/A) 
> CallerIDName= (N/A) 
> ConnectedLineIDNum= 423 
> ConnectedLineIDName=AP-423 
> DNIDDigits= (N/A)
> {code}
> If, before making a call, we delete the information from DNIDDigits, then for PJSIP we get in CallerIDNum=${EXTEN} from which the Dial/Queue is made
> Channel
> Executing [anonym11 at test-line:10] Set("SIP/423-00000038", "CALLERID(dnid)= ") in new stack
> {code}
> Name= SIP/423-00000038 
> Type= SIP 
> UniqueID= 1582896895.436 
> LinkedID= 1582896895.436 
> CallerIDNum= 423 
> CallerIDName= AP-423 
> ConnectedLineIDNum= (N/A) 
> ConnectedLineIDName=(N/A) 
> DNIDDigits= (N/A)
> {code}
> Dstchannel
> {code}
> Name= PJSIP/666-00000006 
> Type= PJSIP 
> UniqueID= 1582896895.439 
> LinkedID= 1582896895.436 
> CallerIDNum= anonym11 
> CallerIDName= (N/A) 
> ConnectedLineIDNum= 423 
> ConnectedLineIDName=AP-423 
> DNIDDigits= (N/A)
> {code}
> Is it really a bug or is it specifically made this way?
> If specifically-how to disable it?



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