[asterisk-users] Having problems with BLF

John john at vetsurgeon.org.uk
Fri Mar 5 09:14:50 CST 2010


Yes- followed all 3 wiki instructions. Thanks for naming tips! Does
this log help at all? Looks like the PBX isn't sending the SIP
messages- I notice the previous NOTIFY messages said (queued)- does
this mean anything?

John

PBX*CLI> sip show subscriptions
Peer             User        Call ID      Extension        Last state
   Type            Mailbox
192.168.13.114   222         3c26707958d  223 at default      Idle
   dialog-info+xml <none>
1 active SIP subscription

My sip trace for 222:
PBX*CLI> set debug peer 222
SIP Debugging Enabled for IP: xx.69.xx.yy:2064
    -- Executing [223 at default:1] SIPAddHeader("SIP/221-09c99e60",
""Alert-Info:<http://nohost>;info=alert-internal;x-line-id=0"") in new
stack
    -- Executing [223 at default:2] Dial("SIP/221-09c99e60",
"SIP/223||tT") in new stack
    -- Called 223
 Extension Changed 223[default] new state Ringing for Notify User 222 (queued)
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager.d/README.conf': Found
  == Manager 'john' logged on from 127.0.0.1
  == Manager 'john' logged off from 127.0.0.1
    -- SIP/223-09ca07c0 is ringing
    -- Got SIP response 603 "Decline" back from xx.xx.xx.xx [THIS IS
DIALLED EXTENSION 223 NOT ACCEPTING CALL]
    -- SIP/223-09ca07c0 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [223 at default:3] Hangup("SIP/221-09c99e60", "") in new stack
  == Spawn extension (default, 223, 3) exited non-zero on 'SIP/221-09c99e60'
 Extension Changed 223[default] new state Idle for Notify User 222 (queued)
PBX*CLI> sip set debug off

On 5 March 2010 14:42, Philipp von Klitzing
<klitzing at pool.informatik.rwth-aachen.de> wrote:
> 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
>
>
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