asterisk users - Mar 2010 - Having problems with BLF

Hi,

I'm having a problem getting a snom 300 to work with BLF (extension
222). I've set it to watch extension 220 in the function key config
pages as per the wiki (BLF, <sip:220 at server.com>) but I can't get
the
light to come on when 220 is ringing. The SIP trace page doesn't show
anything coming from my PBX when 220 is ringing or in use. Any help
much appreciated as this has been driving me mad for the last 2 days!
Is this an asterisk config prob (Asterisk 1.4.21.2)?

John

Console output:
    -- Executing [220 at default:1] SIPAddHeader("SIP/221-08ddaf00",
""Alert-Info:<http://nohost>;info=alert-internal;x-line-id=0"")
in new
stack
    -- Executing [220 at default:2] Dial("SIP/221-08ddaf00",
"SIP/220||tT") in new stack
    -- Called 220
 Extension Changed 220[blf] new state Ringing for Notify User 222 (queued)
    -- SIP/220-08e0b528 is ringing

core show hints
    -= Registered Asterisk Dial Plan Hints =-
                    223 at blf                 : SIP/223
State:Idle            Watchers  0
                    222 at blf                 : SIP/222
State:Idle            Watchers  0
                    221 at blf                 : SIP/221
State:Idle            Watchers  0
                    220 at blf                 : SIP/220
State:Idle            Watchers  1
----------------
- 4 hints registered

sip.conf
[general]
...
allowsubscribe=yes
subscribecontext=blf
notifyringing=yes
notifyhold=yes
call-limit=99
limitonpeers=yes

[220]
type=friend
username=220
secret=xxxxxxxxxxxxx
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833

[221]
type=friend
username=221
secret=xxxxxxxxxxxxxxx
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833

[222]
type=friend
username=222
secret=xxxxxxxxx
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833
mailbox=422
vmexten=702
fromdomain=sip3.xxxxx.co.uk

[223]
type=friend
username=223
secret=xxxxx
host=dynamic
call-limit=3
qualify=yes
nat=yes
dtmfmode=rfc2833

extensions.conf
[default]
include => blf
exten =>
_2XX,1,SIPAddHeader("Alert-Info:<http://nohost>\;info=alert-internal\;x-line-id=0")
exten => _2XX,n,DIAL(SIP/${EXTEN},,tT)
exten => _2XX,n,Hangup

[blf]
exten => 220,hint,SIP/220
exten => 221,hint,SIP/221
exten => 222,hint,SIP/222
exten => 223,hint,SIP/223

Hi!
> I'm having a problem getting a snom 300 to work with BLF (extension
> 222). I've set it to watch extension 220 in the function key config
> pages as per the wiki (BLF, <sip:220 at server.com>) but I can't
get the
> light to come on when 220 is ringing. The SIP trace page doesn't show
> anything coming from my PBX when 220 is ringing or in use.
First try with "Extension" instead of "BLF".
Which Wiki page are you referring to exactly?

For example:
http://www.voip-info.org/wiki-Asterisk+phone+snom
http://wiki.snom.com/Interoperability/PBX/Asterisk
http://wiki.snom.com/Features/Extension_Monitoring

Then do a sip debug on your PBX to see if Asterisk is sending the device 
state information. If it is then you need to check your network setup 
(and make sure 222 is registered to the PBX as you might have instructed 
that phone to refuse SIP messages from anyone else).

SIP SHOW SUBSCRIPTIONS might also reveal some more details.

Also:

- It is not advisable to name your sip peers with 22x = phone numbers. 
Those are devices that deserve device names. These usernames are far too 
easy to guess for a brute force attack, and they will put you into 
trouble when you re-arrange your diaplan.

- Maybe except for [222] you most certainly do not need the
"username="
statements. It does not do what you think it does. ;->

Philipp

Have just tried the asterisk setup on an old server that I placed on
our LAN (i.e. server and extensions all on same subnet). BLF worked as
expected, no "queued" messages. Could NAT be the problem?

John

On 6 March 2010 10:16, John <john at vetsurgeon.org.uk>
wrote:
> added canreinvite=yes
>
>> Do add a "SIP SET DEBUG IP 192.168.13.114" to the game and
see what is
>> happening.
> nothing is sent to that IP address. I then turned on STUN in the Snom
> phone and set nat=no for that phone in sip.conf. The IP shown in
> subscriptions then changed from the private ip to the public WAN IP of
> our router. I debugged that ip/port and saw nothing sent to 222.
> Messages still say (queued)
>
>> What does the result of "route -n" look like?
> all looks fine, internet connectivity/dns all fine
>
>> What does "SIP SHOW PEER 223" say concerning
"Status" and "Addr->IP", can
>> 223 be called?
> 222, 223 + all other extensions say OK (since qualify=yes on), and IP
> is the public WAN IP of our router. All extensions can call each other
> OK
>
> Thanks for help so far
>
> John
>

Hi!
> Have just tried the asterisk setup on an old server that I placed on
> our LAN (i.e. server and extensions all on same subnet). BLF worked as
> expected, no "queued" messages. Could NAT be the problem?
Possibly - this is what chan_sip.c says just before "(queued)":

"We already have a NOTIFY sent that is not answered. Queue the state up.
If many state changes happen meanwhile, we will only send a notification 
of the last one."

But of course there is always the chance to uncover a bug. :-)

You might want to check if your router is somehow "SIP aware", acts as
a
SIP proxy (possibly a transparent proxy), and if yes, try to turn that 
off. And maybe share what routers (or routerS) you employ.

Philipp