Here are some other files. Why asterisk send sip OPTION message to agents ?
Harry //////////////////////////////////////////////////// 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as747a6ef0 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /////////////////////////////////////////////////////// --- harry gaillac <[EMAIL PROTECTED]> a écrit : > 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 > > [email protected] > > > 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> _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > [email protected] > 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 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
