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

[BJ Weschke]

 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
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