I had that alread set in sip_profiles/internal.xml together with these values:

 

<param name="manage-presence" value="true"/>

<param name="send-presence-on-register" value="true"/>

<param name="presence-probe-on-register" value="true"/>

<param name="presence-proto-lookup" value="true"/>

<param name="manage-shared-appearance" value="true"/>

<param name="dbname" value="share_presence"/>

<param name="presence-hosts" value="$${domain}"/>

 

Still no rows in sip_presence

 

 

Von: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Freitag, 26. Oktober 2012 00:46
An: freeswitch-dev@lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] Question about sofia_presence.c

 

err in sofia profile.

 

<param name="multiple-registrations" value="true"/>

On Thu, Oct 25, 2012 at 5:45 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

then enable the multiple-registrations param in your sofia-presence

 

On Thu, Oct 25, 2012 at 1:40 AM, Gerald Weber <gerald.weber@besharp.at> wrote:

So you mean a sip packet like this ?

 

No.     Time           Source                Destination           Protocol Length Info

     74 4.780876000    192.168.20.150        192.168.20.73         SIP/XML  1174   Request: PUBLISH sip:2001@192.168.20.73

 

Frame 74: 1174 bytes on wire (9392 bits), 1174 bytes captured (9392 bits) on interface 0

Ethernet II, Src: HewlettP_33:a8:77 (e8:39:35:33:a8:77), Dst: Vmware_8c:00:1c (00:50:56:8c:00:1c)

Internet Protocol Version 4, Src: 192.168.20.150 (192.168.20.150), Dst: 192.168.20.73 (192.168.20.73)

    Version: 4

    Header length: 20 bytes

    Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00: Not-ECT (Not ECN-Capable Transport))

    Total Length: 1160

    Identification: 0x1de8 (7656)

    Flags: 0x02 (Don't Fragment)

    Fragment offset: 0

    Time to live: 128

    Protocol: UDP (17)

    Header checksum: 0x0000 [incorrect, should be 0x2e4d (may be caused by "IP checksum offload"?)]

    Source: 192.168.20.150 (192.168.20.150)

    Destination: 192.168.20.73 (192.168.20.73)

    [Source GeoIP: Unknown]

    [Destination GeoIP: Unknown]

User Datagram Protocol, Src Port: 16066 (16066), Dst Port: sip (5060)

Session Initiation Protocol

    Request-Line: PUBLISH sip:2001@192.168.20.73 SIP/2.0

    Message Header

    Message Body

        eXtensible Markup Language

            <?xml

            <presence

                xmlns='urn:ietf:params:xml:ns:pidf'

                xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model'

                xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid'

                xmlns:c='urn:ietf:params:xml:ns:pidf:cipid'

                xmlns:lt='urn:ietf:params:xml:ns:location-type'

                xmlns:caps='urn:ietf:params:xml:ns:pidf:caps'

                entity='sip:2001@192.168.20.73'>

                <tuple

                    id='t7301aa0d'>

                    <status>

                        <basic>

                        </status>

                    </tuple>

                <dm:person

                    id='p74c68736'>

                    <rpid:activities>

                        <rpid:busy/>

                        </rpid:activities>

                    <dm:note>

                        Do not disturb

                        </dm:note>

                    </dm:person>

                </presence>

 

 

This i sent from a bria3 client to fs but i dont get any entry in sip_presence because of the check in line 4204.

 

 

Von: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 24. Oktober 2012 20:01
An: freeswitch-dev@lists.freeswitch.org
Betreff: Re: [Freeswitch-dev] Question about sofia_presence.c

 

it has to send a publish to update sip_presence using the pidf method.

 

On Wed, Oct 24, 2012 at 10:55 AM, Gerald Weber <gerald.weber@besharp.at> wrote:

Hi all,

 

i’m currently extending my list_users command to show the presence status of registered sip devices.

I have a snom 360 and a bria softphone here to test with, calling one from the other shows 2 rows in sip_dialogs table so presence

is basically working.

 

Both phones have the ability to set the presence to, e.g. busy or meeting or whatever.

 

The only problem i have is, that there is no line in sip_presence table when i change the presence on the phone (or using presence in)

Digging through the code, i found that this is  because of the if statement in src/mod/endpoints/mod_sofia/sofia_presence.c line 4204

Looks like a check for multiple registrations, but how is this intended to work with only one registration per user ?

I changed the if to if (count > 0)  for testing and i can see the correct entries in sip_registrations.

 

I’m not quite sure if i understand the purpose of this code so i didn’t raise a jira.

Is this a bug ? If not, does the sip_presence table only get filled when there are more than one registration per user ?

 

Thanks

gw

 

 

 

 

 


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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev@lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org



 

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900



 

--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900