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

[harry gaillac]

Hello,

Asterisk don't support IM presence because of no proxy
function in chan_sip !

Regards
Harry

--- harry gaillac <gaillacharry at yahoo.fr> a écrit :

> When the polycom ip300 phone (1.6.2) send
> registration
> 
> SUBSCRIBE message is sent to buddies from directory
> file so NOTIFY is received from these one.
> 
> When I want to change status the ip phone don't send
> NOTIFY to subscriber unlike SER which is a proxy!!!
> Why?
> 
> Harry
> --- harry gaillac <gaillacharry at yahoo.fr> a écrit :
> 
> > 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
> 
=== message truncated ===



	

	
		
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