[Asterisk-Users] IM / presence asterisk-1.2-RC1

[harry gaillac]

Sorry,

Here are some files 

Harry
--- BJ Weschke <bweschke at gmail.com> a écrit :

>  This is good debugging info you've listed below,
> but this isn't a sip
> debug/trace.
> 
>  To do that, first verify in your logger.conf file
> you have the following line:
> 
>  full => notice,warning,error,debug,verbose
> 
>  Then, if you needed to add anything to logger.conf,
> please first
> restart Asterisk so those new settings take effect.
> 
>  Then, from the CLI issue "set verbose 5" and "set
> debug 5" and
> finally "sip debug".
> 
>  The repeat your dialing steps.
> 
>  The sip debug/trace will then be contained in
> /var/log/asterisk/full
> if /var/log/asterisk is where your log files are
> kept.
> 
>  With that, we can have a better idea of what's
> happening/not
> happening to give you the issue you're having.
> 
> 
> On 11/10/05, harry gaillac <gaillacharry at yahoo.fr>
> wrote:
> > I did it !?
> >
>
//////////////////////////////////////////////////////
> > Connected to Asterisk 1.2.0-rc1 currently running
> on
> > serveur1 (pid = 1125)
> > Verbosity is at least 4
> > serveur1*CLI> sip show subscriptions
> > Peer             User        Call ID     
> Extension
> >    Last state     Type
> > 192.168.0.21     86          f1682d8d-8f  84
> >    Idle           xpidf+xml
> > 192.168.0.21     86          5f32aec-95b  85
> >    Idle           xpidf+xml
> > 192.168.0.20     84          cb424ae1-e4  86
> >    Idle           xpidf+xml
> > 192.168.0.20     84          715fac66-a9  87
> >    Idle           xpidf+xml
> > 4 active SIP subscriptions
> > serveur1*CLI>
> >
>
//////////////////////////////////////////////////////
> > serveur1*CLI> sip show peers
> > Name/username              Host            Dyn Nat
> ACL
> > Port     Status
> > 87/87                      192.168.0.21     D   N
> > 5060     OK (84 ms)
> > 86/86                      192.168.0.21     D   N
> > 5060     OK (97 ms)
> > 85/85                      192.168.0.20     D   N
> > 5060     OK (87 ms)
> > 84/84                      192.168.0.20     D   N
> > 5060     OK (96 ms)
> > 4 sip peers [4 online , 0 offline]
> > serveur1*CLI>
> >
>
///////////////////////////////////////////////////////
> > my sip.conf:
> > [general]
> > context=local                   ; Default context
> for incoming calls
> >                                ; if asterisk was
> compiled with OSP support.
> > realm=nxs.yi.org                ; Realm for digest
> authentication
> >                                ; defaults to
> "asterisk"
> >                                ; Realms MUST be
> globally unique according to RFC
> > 3261
> >                                ; Set this to your
> host name or domain name
> > bindport=5060                   ; UDP Port to bind
> to (SIP standard
> > port is 5060)
> > bindaddr=nxs.yi.org             ; IP address to
> bind to (0.0.0.0
> > binds to all)
> > srvlookup=yes                   ; Enable DNS SRV
> lookups on outbound
> > calls
> > tos=lowdelay                    ;
> > lowdelay,throughput,reliability,mincost,none
> > maxexpirey=3600                 ; Max length of
> incoming
> > registration we allow
> > defaultexpirey=1000             ; Default length
> of
> > incoming/outoing registration
> > allow=all                       ; First disallow
> all codecs
> > musicclass=default              ; Sets the default
> music on hold
> > class for all SIP calls
> > language=fr                     ; Default language
> setting for all
> > users/peers
> > rtptimeout=60                   ; Terminate call
> if 60 seconds of no
> > RTP activity
> > tpholdtimeout=300               ; Terminate call
> if 300 seconds of
> > no RTP activity
> > useragent=Asterisk PBX          ; Allows you to
> change the
> > user agent string
> > dtmfmode = rfc2833              ; Set default
> dtmfmode for sending
> > DTMF. Default: rfc2833
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
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