[Asterisk-Users] IM / presence asterisk-1.2-RC1
Sergey Okhapkin
sos at sokhapkin.dyndns.org
Fri Nov 11 07:53:00 MST 2005
Asterisk sends OPTIONS message if the device have "qualify=NNN" option
set.
On Fri, 2005-11-11 at 11:24 +0100, harry gaillac wrote:
> 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:84 at 192.168.0.20 SIP/2.0
> Via: SIP/2.0/UDP
> 80.119.11.222:5060;branch=z9hG4bK4a119599;rport
> From: "asterisk"
> <sip:asterisk at 80.119.11.222>;tag=as747a6ef0
> To: <sip:84 at 192.168.0.20>
> Contact: <sip:asterisk at 80.119.11.222>
> Call-ID:
> 0be39a0e4bdea3802b7386bb60009605 at 80.119.11.222
> 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 <gaillacharry at yahoo.fr> a écrit :
>
> > 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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> > >
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> >
> >
> >
> >
> >
> >
> ___________________________________________________________________________
> >
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>
>
>
> ___________________________________________________________________________
> 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 --
>
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