[Asterisk-Users] storing cdr in two databases
Hi, Is it possible to send CDR to a database (cdr_mysql.so for example) and to files (cdr_csv.so) ? As soon as I activated CDR writes to mysql, Master.csv stopped to grow, and since CDRs seems to be registered in a linked list in cdr.c I thought it was possible... TIA, -- Ludovic DROLEZ Linbox / FreeALter Soft 152 rue de Grigy - Technopole Metz 2000 57070 METZ tel : 03 87 50 87 90fax : 03 87 75 19 26 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call waiting trouble with 7912 cisco phones
Hello ! We have 7912G SIP phones with the 1.02.00 firmware. *Sometimes* when you call someone who is already on the phone, our PBX receives immediatly a 302 Moved Temporarily SIP message, so that the 2nd caller is forwarded to the voicemail instead of waiting 20s (Allow Call Waiting is set to 1, and Forward to VMail Delay to 20). Since I know that Cisco, won't fix the bug before 20 years, I wondered if I can find a work-around thanks to asterisk. As you see in the following trace: 1- The incoming call is 'broadcast' (SIP/8791SIP/8792SIP/8793) 2- Phones are ringing except one, which sends a 302 message 3- Asterisk immediatly redirects the incoming call to the voicemail = TRACE -- Executing Dial(CAPI[contr2/387508790]/116, SIP/8791SIP/8792SIP/8793SIP/8794SIP/8795SIP/8797SIP/8798|20|t) in new stack -- Called 8791 -- Called 8792 -- Called 8793 -- Called 8794 -- Called 8797 -- Called 8798 -- Got SIP response 302 Moved Temporarily back from 192.168.0.202 -- Now forwarding CAPI[contr2/387508790]/116 to '[EMAIL PROTECTED]' (thanks to SIP/8792-b8bc) -- Executing Wait(Local/[EMAIL PROTECTED],2, 1) in new stack -- SIP/8797-b246 is ringing -- SIP/8791-2583 is ringing -- SIP/8793-e1eb is ringing -- SIP/8798-cce6 is ringing -- SIP/8794-3c01 is ringing -- Executing VoiceMailMain2(Local/[EMAIL PROTECTED],2, 06XXX) in new stack Is there a way to tell asterisk to ignore 302 messages when a call is broadcast (A nice Dail option) ? TIA, -- Ludovic DROLEZ Linbox / FreeALter Soft 152 rue de Grigy - Technopole Metz 2000 57070 METZ tel : 03 87 50 87 90fax : 03 87 75 19 26 ___ 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
Re: [Asterisk-Users] Call waiting trouble with 7912 cisco phones
Philipp von Klitzing wrote: How about this pseudo code: [default] 1,Dial(Sip/1Sip/2) 2,SetVar(foo=x) 3,Goto(international,8500,1) 102,SetVar(foo=x) 103,Goto(international,8500,1) [international] 8500,1,GotoIf(foo=x THEN voicemail ELSE callotherphones) Many thanks for the reply, but with 'callotherphones' I think that there would be a loop or a '302 message storm'... Do you know if I can replace a 'callotherphones' by a 'do nothing, continue ringing other phones' ? How could I code that ? Cheers, -- Ludovic DROLEZ Linbox / FreeALter Soft 152 rue de Grigy - Technopole Metz 2000 57070 METZ tel : 03 87 50 87 90fax : 03 87 75 19 26 ___ 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
[Asterisk-Users] Cisco 7912 speed dials
Hi ! We have Cisco 7912 phones, and the doc says that I can create up to four speed dial buttons on my phone using the Cisco CallManager. Does anyone knows which protocol is used to configure speed dials (Is it documented somewhere) ? Did someone tried to reverse engineer the protocol ? It would be cool, not having to pay $15000 just for configuring speed dials on those phones ;-D Cheers, -- Ludovic Drolez. http://www.palmopensource.com - The PalmOS Open Source Portal http://www.drolez.com - Personal site - Linux and PalmOS stuff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing calls for a fancy address book app
Hi ! I'd like to connect phpgroupware to asterisk: when a user click on a phone number, his phone rings and he gets connected to the number he just clicked. I've tried by putting various files in /var/spool/asterisk/outgoing, without results (we are using SIP phones + CAPI channels). Is there a way to do that ? (If it's impossible (something impossible in *, LOL ?!?) I will create an extension which the phpgroupware user should call, then it would DBGet the number my app has DBPut when the user clicked on the number, and call it.) Cheers, -- Ludovic Drolez. http://www.palmopensource.com - The PalmOS Open Source Portal http://www.drolez.com - Personal site - Linux and PalmOS stuff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
