Hi Mustafa N. Deeb, Monday, February 7, 2005, 3:58:08 PM, Вы писали:
===8<==============Original message text=============== Mustafa N. Deeb> Hi Mustafa N. Deeb> Has anyone got the 7902 phone work with asterisk , the only thing I was able Mustafa N. Deeb> to do with it, is to dial from it.. Mustafa N. Deeb> It doesn't ring ,and if you pick the handset for 30 secs , asterisk crashes. Mustafa N. Deeb> I know cisco is not planning on releasing a SIP image for it , so we are Mustafa N. Deeb> stuck with SCCP. Mustafa N. Deeb> Regards Mustafa N. Deeb> -----Original Message----- Mustafa N. Deeb> From: [EMAIL PROTECTED] Mustafa N. Deeb> [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Mustafa N. Deeb> Sent: Monday, February 07, 2005 1:22 AM Mustafa N. Deeb> To: Asterisk Users Mailing List - Non-Commercial Discussion Mustafa N. Deeb> Subject: Re: [Asterisk-Users] Encrypted VOIP? Mustafa N. Deeb> Wolfgang S. Rupprecht wrote: >>In theory, the Sipura line supports SRTP. I've got both a spa-841 and >>a spa-3000 that have config areas for loading the srtp rsa keys. >>Unfortunately there isn't enough information given by sipura as to how >>to generate these rsa keys. (eg. can one use an openssl generated >>key?) >> >> Mustafa N. Deeb> In the Sipura support area (authentication required), there is a tool to Mustafa N. Deeb> generate Mini Certificates for this. Mustafa N. Deeb> _______________________________________________ Mustafa N. Deeb> Asterisk-Users mailing list Mustafa N. Deeb> [email protected] Mustafa N. Deeb> http://lists.digium.com/mailman/listinfo/asterisk-users Mustafa N. Deeb> To UNSUBSCRIBE or update options visit: Mustafa N. Deeb> http://lists.digium.com/mailman/listinfo/asterisk-users Mustafa N. Deeb> _______________________________________________ Mustafa N. Deeb> Asterisk-Users mailing list Mustafa N. Deeb> [email protected] Mustafa N. Deeb> http://lists.digium.com/mailman/listinfo/asterisk-users Mustafa N. Deeb> To UNSUBSCRIBE or update options visit: Mustafa N. Deeb> http://lists.digium.com/mailman/listinfo/asterisk-users ===8<===========End of original message text=========== Try do this. Install chan_sccp (Asterisk SCCP2 channel) from hear http://chan-sccp.sourceforge.net/. In sccp.conf wright: [general] ; How often the SCCP device does a keepalive ping ; Default: 5 seconds keepalive = 5 ; default context that will be used if nothing else is specified for ; a particular device/line context = default dateFormat = D-M-Y ; M-D-Y in any order (5 chars max) bindaddr = 0.0.0.0 ; replace 1.2.3.4 with the ip address of the ; asterisk box. port = 2000 ; listen on port 2000 (Skinny, default) ; ; Typical config for a 7902 [SEPXXXXXXXXXXXX] ; Device name (SEP+Device Mac-address) type = 7902 ; Offical identifier description = CISCO CP-7902G autologin = cisco imgversion = 031023A [cisco] id = 1801 Label = Cisco description = Cisco CP-7902G context = default ; Or not default callwaiting = 1 mailbox = 1801 callerid = "Cisco IP-Phone" <1801> In extentions.conf i do: [sccp] exten => 1801,1,SetCalledParty("CISCO IP-Phone" <1801>) exten => 1801,2,Wait,1 exten => 1801,3,Answer exten => 1801,4,NoOp("Call for "${EXTEN}) exten => 1801,5,Dial(SCCP/cisco,60) exten => 1801,6,VoiceMail(u${EXTEN}) exten => 1801,7,Congestion exten => 1801,100,Busy Phone is work fine. But i'm have a one problem. When i place a call from 7902G to any across Asterisk, after i hear RingOut need press a hold button twice. And we can speak with the caused subscriber. Otherwise in the handset i hear RingOut. -- Best regards, Andrew A. Kochetkoff mailto:[EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
