Sorry, Here are some files
Harry --- BJ Weschke <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > 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/ > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com
messages.serveur1.home.net
Description: 1676272990-messages.serveur1.home.net
debug.serveur1.home.net
Description: 3484436676-debug.serveur1.home.net
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