[Asterisk-Users] IM / presence asterisk-1.2-RC1
harry gaillac
gaillacharry at yahoo.fr
Thu Nov 10 05:53:27 MST 2005
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
promiscredir = no ; If yes, allows 302
or REDIR to non-local SIP address
nat=yes
qualify=500
[84]
type=friend
secret=84
context=local
host=dynamic
mailbox=84
allow=all
[85]
type=friend
secret=85
context=local
host=dynamic
mailbox=85
allow=all
[86]
type=friend
secret=86
context=local
host=dynamic
mailbox=86
allow=all
[87]
type=friend
secret=87
context=local
host=dynamic
mailbox=87
allow=all
//////////////////////////////////////////////////////
my extension.conf
;
[general]
;
static=yes
writeprotect=no
switch => Realtime/mailbox at extensions
;
[globals]
;
[local]
exten => 80,1,Answer
exten => 80,2,Dial(Zap/g2,14)
exten => 80,3,VoiceMail(u80)
exten => 80,103,VoiceMail(b80)
exten => 84,hint,Sip/84
exten => 84,1,Answer
exten => 84,2,Dial(Sip/84,10)
exten => 84,3,VoiceMail(u84)
exten => 84,103,VoiceMail(b84)
exten => 85,hint,Sip/85
exten => 85,1,Answer
exten => 85,2,Dial(Sip/85,10)
exten => 85,3,VoiceMail(u85)
exten => 85,103,VoiceMail(b85)
exten => 86,hint,Sip/86
exten => 86,1,Answer
exten => 86,2,Dial(Sip/86,10)
exten => 86,3,VoiceMail(u86)
exten => 86,103,VoiceMail(b86)
exten => 87,hint,Sip/87
exten => 87,1,Answer
exten => 87,2,Dial(Sip/87,10)
exten => 87,3,VoiceMail(u87)
exten => 87,103,VoiceMail(b87)
include => mailbox
include => apps
include => pstn
[mailbox]
exten => 700,1,VoiceMailMain()
[pstn]
exten => s,1,Answer
exten => s,2,Goto(local,84,1)
include => outgoing-pstn
[outgoing-pstn]
ingnorepat => 0
exten => _0XXXX,1,Dial(Zap/g1/${EXTEN:1})
exten => _0XXXX.,1,Dial(Zap/g1/${EXTEN:1})
exten => _0XXXX.,3,Hangup
//////////////////////////////////////////////////////
Regards
Harry
--- BJ Weschke <bweschke at gmail.com> a écrit :
> Harry,
>
> The monitoring of buddies on Polycom phones is
> possible with the
> release candidate for v1.2. We've asked for a sip
> debug/trace from you
> to try and troubleshoot your problem, and you
> haven't provided that to
> date.
>
> On 11/10/05, harry gaillac <gaillacharry at yahoo.fr>
> wrote:
> > Hello,
> >
> > Does asterisk's team will improve IM and presence
> in
> > asterisk-1.2 !
> >
> > Send Sip MESSAGE is impossible.
> > When the buddies status change nothing is
> happened.
> >
> > How asterisk's team plan to solve this problem ?
> >
> > Regards
> > Harry
> >
> >
> >
> >
> >
> >
> >
>
___________________________________________________________________________
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