[asterisk-bugs] [JIRA] (ASTERISK-17650) Remote-Party ID not added when CALLERID(num)=<empty>

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-17650?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-17650:
-----------------------------------

    Assignee: Michaël Arnauts
      Status: Waiting for Feedback  (was: Open)

Is this still a problem under a current supported version of Asterisk?

> Remote-Party ID not added when CALLERID(num)=<empty>
> ----------------------------------------------------
>
>                 Key: ASTERISK-17650
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-17650
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.3
>            Reporter: Michaël Arnauts
>            Assignee: Michaël Arnauts
>            Severity: Minor
>
> i've configured in my sip.conf to send an rpid with sendrpid=yes (also tried with sendrpid=rpid and even sendrpid=pai), but my SIP message is not containing a Remote-Party-Id. It was correctly working in 1.6.2...
> I can provide a packet dump if you like...
> ****** ADDITIONAL INFORMATION ******
> This is a anonymous call without a number to the sip account dest.
> The privacy:id is added, but i don't see the Remote-Party-Id
> voice-trunk-ix*CLI> sip show peer dest 
>   * Name       : dest
>   ...
>   Trust RPID   : No
>   Send RPID    : Yes
> Call trace:
>     -- AGI Script Executing Application: (Set) Options: (CALLERPRES()=prohib_not_screened)
>     -- AGI Script Executing Application: (SIPAddHeader) Options: (Privacy:id)
>     -- AGI Script Executing Application: (Set) Options: (CALLERID(ANI)=)
>     -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=)
>     -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Unknown)
>     -- <SIP/source-000001f4>AGI Script set-callerid-outbound.agi completed, returning 0
>     -- Executing [+3224019700 at to-external:3] ExecIf("SIP/source-000001f4", "0?AGI(record-call.agi)") in new stack
>     -- Executing [+3224019700 at to-external:4] NoOp("SIP/source-000001f4", "Outbound call: Number  dailed to +3224019700") in new stack
>     -- Executing [+3224019700 at to-external:5] Dial("SIP/source-000001f4", "SIP/dest/+3224019700") in new stack
>   == Using UDPTL CoS mark 5
>   == Using SIP RTP CoS mark 5
> Audio is at 5060
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Reliably Transmitting (no NAT) to x.x.x.129:5060:
> INVITE sip:+3224019756 at x.x.x.129 SIP/2.0
> Via: SIP/2.0/UDP x.x.x.3:5060;branch=z9hG4bK4d855e44
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at x.x.x.3>;tag=as7d5b7e08
> To: <sip:+3224019756 at x.x.x.129>
> Contact: <sip:asterisk at x.x.x.3:5060>
> Call-ID: 58d9718b7e6f60c345282466034d499e at x.x.x.3:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.8.3.2
> Date: Wed, 06 Apr 2011 13:30:02 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Privacy: id
> Content-Type: application/sdp
> Content-Length: 221
> v=0
> o=root 2099576720 2099576720 IN IP4 x.x.x.3
> s=Asterisk PBX 1.8.3.2
> c=IN IP4 x.x.x.3
> t=0 0
> m=audio 14166 RTP/AVP 0 8
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv



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