[Asterisk-Users] IM / presence asterisk-1.2-RC1
harry gaillac
gaillacharry at yahoo.fr
Fri Nov 11 03:24:24 MST 2005
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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> >
> >
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> >
>
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>
>
>
>
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