[OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

Thu Aug 2 13:13:11 EDT 2018

The free Bria client is capable of being a presence user agent. I believe Zoiper will do so as well.

The error response you are getting is 489 Bad Event. Looking at the Event header in the message you are sending from SIPp you have set “E_PRESENCE_PUBLISH”. That is not a defined event package for SIP; that is the name of the internal OpenSIPS event corresponding to presence publishing. Per RFC 3856 the event package for presence is “presence”.

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Giovanni Maruzzelli <gmaruzz at gmail.com>
Reply-To: "gmaruzz at opentelecom.it" <gmaruzz at opentelecom.it>, OpenSIPS users mailling list <users at lists.opensips.org>
Date: Thursday, August 2, 2018 at 1:02 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Ordinary presence server functions of OpenSIPS

study, Fatma, study :)

btw, if you can't "find a softphone capable of being a presence user agent" you may be in the wrong field of studies.

-giovanni

On 2 August 2018 at 15:52, Fatma Raissi <raissifatma at gmail.com<mailto:raissifatma at gmail.com>> wrote:
Good morning,

Thanks again for your answer.
But I can't find a softphone capable of being a presence user agent.
Plus the presence information I need to publish is one variable which is "workload" of the machine.

Here is the SIP message I am using and joined the configuration file. Maybe you can Identify the problem. Thanks

test.xml

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="Basic Message">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id]
keyword.                -->
  <!-- https://en.wikipedia.org/wiki/List_of_SIP_request_methods -->
  <!-- https://www.ietf.org/rfc/rfc3428.txt -->
  <!--
      Content-Type: application/sdp
  -->
  <send retrans="1">
    <![CDATA[
      SUBSCRIBE sip:127.0.0.1:5060<http://127.0.0.1:5060> SIP/2.0.
      Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bKnashds7
      To: sip:127.0.0.1:5060<http://127.0.0.1:5060>
      From: sip:127.0.0.1:5060;tag=12341234
      Call-ID: 12345678x at 127.0.0.1:5060<http://12345678x@127.0.0.1:5060>
      CSeq: 1 SUBSCRIBE
      Max-Forwards: 70
      Expires: 3600
      Event: E_PRESENCE_PUBLISH
      Content_Type: application/pidf+xml
      Contact: sip:127.0.0.1:5060<http://127.0.0.1:5060>
      Content-Length: 5

    ]]>
  </send>
<recv request="MESSAGE|PUBLISH|SUBSCRIBE" crlf="true" regexp_match="true">
  </scenario>

Reponse: 489 Bad event

Aug  2 06:49:44 [40701] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE>
Aug  2 06:49:44 [40701] DBG:maxfwd:is_maxfwd_present: value = 70
Aug  2 06:49:44 [40701] DBG:uri:has_totag: no totag
Aug  2 06:49:44 [40701] DBG:core:parse_headers: flags=78
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: start searching: hash=22792, isACK=0
Aug  2 06:49:44 [40701] DBG:tm:matching_3261: RFC3261 transaction matched, tid=nashds7
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: REF_UNSAFE:[0x7f73f6c49708] after is 1
Aug  2 06:49:44 [40701] DBG:tm:t_lookup_request: transaction found (T=0x7f73f6c49708)
Aug  2 06:49:44 [40701] DBG:tm:t_retransmit_reply: buf=0x7f73f644f600: SIP/2.0 4..., shmem=0x7f73f6c4c678: SIP/2.0 4
Aug  2 06:49:44 [40701] DBG:tm:t_check_trans: UNREF_UNSAFE: [0x7f73f6c49708] after is 0
Aug  2 06:49:44 [40701] DBG:core:destroy_avp_list: destroying list (nil)
Aug  2 06:49:44 [40701] DBG:core:receive_msg: cleaning up
Aug  2 06:49:44 [40700] DBG:core:parse_msg: SIP Request:
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  method:  <SUBSCRIBE>
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  uri:     <sip:127.0.0.1:5060<http://127.0.0.1:5060>>
Aug  2 06:49:44 [40700] DBG:core:parse_msg:  version: <SIP/2.0.>
Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=2
Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=6
Aug  2 06:49:44 [40700] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKnashds7>; state=16
Aug  2 06:49:44 [40700] DBG:core:parse_via: end of header reached, state=5
Aug  2 06:49:44 [40700] DBG:core:parse_headers: via found, flags=2
Aug  2 06:49:44 [40700] DBG:core:parse_headers: this is the first via
Aug  2 06:49:44 [40700] DBG:core:receive_msg: After parse_msg...
Aug  2 06:49:44 [40700] DBG:core:receive_msg: preparing to run routing scripts...
Aug  2 06:49:44 [40700] DBG:core:parse_headers: flags=100
Aug  2 06:49:44 [40700] DBG:core:_parse_to: end of header reached, state=9
Aug  2 06:49:44 [40700] DBG:core:_parse_to: display={}, ruri={sip:127.0.0.1:5060<http://127.0.0.1:5060>}
Aug  2 06:49:44 [40700] DBG:core:get_hdr_field: <To> [20]; uri=[sip:127.0.0.1:5060<http://127.0.0.1:5060>]
Aug  2 06:49:44 [40700] DBG:core:get_hdr_field: to body [sip:127.0.0.1:5060<http://127.0.0.1:5060>

Error! Filename not specified.

Fatma RAISSI  - ENIT Junior Entreprise

Élève ingénieur en télécommunication
Membre d'honneur
Vice-Présidente du mandat 2016-2017
 Tel: (+216) 53 411 311 | Email: raissifatma at gmail.com<mailto:prenom.nom at gmail.com>

2018-08-02 13:48 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>>:
Be ause they have working presence client embedded, and you seems not be able to model it in sipp.

Start with something known to work, softphones, trace the sip messages, then (if needed) do the sipp xml modelization.

-giovanni

On Thu, Aug 2, 2018, 13:45 Fatma Raissi <raissifatma at gmail.com<mailto:raissifatma at gmail.com>> wrote:
Good morning Sir,

Thank you a lot for your answer.
But could you explain why would I use softphones while I have nothing to do with voice or voice over IP.

Cordially,

Error! Filename not specified.

Fatma RAISSI  - ENIT Junior Entreprise

Élève ingénieur en télécommunication
Membre d'honneur
Vice-Présidente du mandat 2016-2017
 Tel: (+216) 53 411 311 | Email: raissifatma at gmail.com<mailto:prenom.nom at gmail.com>

2018-08-02 10:34 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com<mailto:gmaruzz at gmail.com>>:
Use softphones instead of sipp

On Wed, Aug 1, 2018, 12:01 Fatma Raissi <raissifatma at gmail.com<mailto:raissifatma at gmail.com>> wrote:
Good morning Everyone,

I am using OpenSIPS as presence server. I need it just to accomplish very basic and simple presence server functions.

Here is the purpose of my work:

I have 3 machines P, A, B and C.

1) P is the machine in which I have installed OpenSIPS, thus the presence server
2) I want A and B to be the presentities and thus publishing its own presence information (that I precise) into the presence server.
3) I want OpenSIPS to update the watcher C each time there is a change in A or B presence information.

What I have done so far is that I am using the presence server config file that I found here :

https://www.opensips.org/Documentation/Tutorials-Presence-SimplePresConfig

and added all the parameters I found here

http://www.opensips.org/html/docs/modules/2.1.x/presence.html

I am using SIPp in machine A and B  and C to send publish and subscribe messages but I keep getting no answer at all from OpenSIPS although I am sure he is receiving the messages.
I think I am not using the right syntax of SIP messages.

Can you please help me by sending to me an example of SIP subscribe and SIP publish message that I can send from SIPp to OpenSIPS.

- SIP Subscribe message : in which I can precise the list of IP addresses that the machine C needs to know their updates
- SIP Publish message: in which I can precise the presence information I want A and B to publish.

Tell me if you need my config file or the SIP message I am using ( I am pretty sure it is wrong though...)

An other question please: Is it normal that I couldn't find pua.so and rls.so in my modules file?

I really appreciate the help. I couldn't find any example else where.

Cordially,

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--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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