VoIP Mailing List Archives

[fons.vanderbeek at 84-...]

Hello all,

I know it was on the list before but i have some questions about the
Kirk IP600v3, the requested configuration files were send private i guess

Does anybody have the correct SIP settings for handsets connected to the
Kirk. IP600v3

I am particulair intrested in settings regarding:
-Voice Mailbox
-Call waiting
-DTMF settings for e.g. parking an extension with asterisk functionality

Lately i'am having also trouble when i initiated a transfer, i can't
take back the call
by pressing "R".

Does anybody use relaxdtmf and or special DTMF timings for correct usage
of the kirk 600v3 ?????
I am using asterisk 1.4.14 and the newest firmware of the Kirk (07-60663 )

When enabling all advanced features of the kirk 600v3 occasionaly
handsets get disconnected, still trying to figure out which of those
features create this disconnection.
When using no features connection to all handsets are stable.

Futhermore i am getting an error on my CLI
Incoming call: Got SIP response 400 "Bad Request" back from 10.0.0.XX

when looking in set debug ip to my wireless server
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 10.0.0.70:5060;branch=z9hG4bK42bce6b1;rport
From: "XXXXXX" <sip:XXXXXXXX at 10.0.0.70>;tag=as7a91af96
To: <sip:240 at 10.0.0.71:16406;user=phone>;tag=2870354154
<http://www.snapanumber.com/>
Call-ID: 35703f0237eed17d74acac2d7ed5d8b1 at 10.0.0.70
CSeq: 14806 BYE
Server: (KIRK Wireless Server 600v3/6.00 dvl-sr2 [07-60663])

XXXXXX = caller id of Calling party
It looks OK, but is giving a Bad request

Does anybody know how to avoid/solve this error, i get a lot of
them........................


Sip.conf for a particulair handset
[235]
type=friend
username = 235
callerid="R Vermeeren mobiel" <235>
host = dynamic
secret = 235
context = default
qualify = yes
login = 235
callgroup = 3
pickupgroup = 3
disallow = all
allow = alaw
call-limit = 6

default section of sip.conf
[general]
dtmfmode=rfc2833
rfc2833compensate=yes
notifyringing=yes
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support.
(Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
; bindport is the local UDP port that
Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
allowsubscribe=yes ; Disable support for subscriptions.
(Default is yes)
subscribecontext = default ; Set a specific context for SUBSCRIBE
requests
notifyringing = yes ; Notify subscriptions on RINGING state
(default: no)
notifyhold = yes ; Notify subscriptions on HOLD state
(default: no)
limitonpeers = yes ; Apply call limits on peers only. This
will improve
useclientcode=yes

When more information is needed, please ask..............