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:[EMAIL PROTECTED]>;tag=as7a91af96 To: <sip:[EMAIL PROTECTED]:16406;user=phone>;tag=2870354154 <http://www.snapanumber.com/> Call-ID: [EMAIL PROTECTED] 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.............. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users