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Summary:ASTERISK-17650: Remote-Party ID not added when CALLERID(num)=<empty>
Reporter:Michaël Arnauts (michaelarnauts)Labels:
Date Opened:2011-04-06 08:35:16Date Closed:2018-01-02 08:44:21.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.3 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description: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@to-external:3] ExecIf("SIP/source-000001f4", "0?AGI(record-call.agi)") in new stack
   -- Executing [+3224019700@to-external:4] NoOp("SIP/source-000001f4", "Outbound call: Number  dailed to +3224019700") in new stack
   -- Executing [+3224019700@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@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@x.x.x.3>;tag=as7d5b7e08
To: <sip:+3224019756@x.x.x.129>
Contact: <sip:asterisk@x.x.x.3:5060>
Call-ID: 58d9718b7e6f60c345282466034d499e@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
Comments:By: Andrew Latham (lathama) 2011-04-06 11:41:17

https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
By: Michaël Arnauts (michaelarnauts) 2011-04-07 02:15:17

I've narrowed the issue down to this:
    -- AGI Script Executing Application: (Set) Options: (CALLERID(num)=)

When you set the number to <empty>, the Remote Party Id is not added, and the prohib_not_screened is therefore not passed on to the next hop. This results in the next hop displaying "asterisk" as a callerid, while it should be hidden.
By: Joshua C. Colp (jcolp) 2017-12-19 06:57:55.325-0600

Is this still a problem under a current supported version of Asterisk?
By: Asterisk Team (asteriskteam) 2018-01-02 08:44:21.302-0600

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines