Danny, Why would you think this is a "circumvent"? I'm using a nice feature of 1.8 where I can create any CDR field I like and populate it by using the CDR(<fieldname>) function. While all other fields that I created are populated properly (however before the 'dial' commences) it seems like at this point of the dial plan the CDR is closed for editing even though I configured endbeforehexten=no in my cdr.conf.
It might be related to issue ASTERISK-18875 as suggested by Daniel (Vol. 89 issue 8, topic 9). I'll be happy to know if someone has a different knowledge on the subject, otherwise I'll simply follow ASTERISK-18875. My problem with this issue is that it is defined as low importance which means that it will probably take long to handle if at all... Harel ************************************************************** Message: 4 Date: Tue, 6 Dec 2011 07:29:54 -0600 From: "Danny Nicholas" <[email protected]> Subject: Re: [asterisk-users] Populate CDR issues To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[email protected]> Message-ID: <[email protected]> Content-Type: text/plain; charset="us-ascii" IMO you are trying to circumvent basic Asterisk functionality. It's your CDR so you can do what you want with it - I think the answer to this is to populate another DB with the live call data, then update the CDR from that after the call has ended (perhaps a daemon). From: [email protected] [mailto:[email protected]] On Behalf Of Harel Cohen Sent: Tuesday, December 06, 2011 3:16 AM To: [email protected] Subject: [asterisk-users] Populate CDR issues Hello Everyone, I didn't get a reply to my problem below so I'm posting again just in case someone who might be able to help missed my previous post. Thank You. **************************************************************************** ********* Hello list, I'm trying to populate my CDR logs with values which are available after the call has started (e.g. signalling IP of remote user, media IP, codec etc.). While CHANNEL function give me all I need for the incoming leg (leg A), I can't get the relevant values for the outgoing channel. I've tried using the option 'U' with my dial command (execute subroutine for called channel after called channel answered but before the call is bridged). While this throws the correct information to the console it does not populate the CDRs accordingly. Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC and the table therein contains the relevant fields. This is the console with 'very-verbose' output for the 'Dial' application where office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP 192.168.20.226. My comments added prefixed by ** and on separate line: ** channel here is source channel: SIP/office_Admin2-00000015 [Dec 1 12:14:31] -- Executing [316@InternalDP:5] Dial("SIP/office_Admin2-00000015", "SIP/office_ServerRoom,,FgU(jump2SetVar)") in new stack [Dec 1 12:14:31] == Using UDPTL CoS mark 5 [Dec 1 12:14:31] == Using SIP RTP CoS mark 5 [Dec 1 12:14:31] -- Called SIP/office_ServerRoom [Dec 1 12:14:31] -- SIP/office_ServerRoom-00000016 is ringing [Dec 1 12:14:31] -- SIP/office_ServerRoom-00000016 is ringing [Dec 1 12:14:33] -- SIP/office_ServerRoom-00000016 answered SIP/office_Admin2-00000015 ** from here the channel is the destination channel: SIP/office_ServerRoom-00000016 [Dec 1 12:14:33] -- Executing [s@jump2SetVar:1] Gosub("SIP/office_ServerRoom-00000016", "SetVar,postdial,1") in new stack ** This is how I obtain channel information: ** exten => postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peer name)},port)}) ; resulting format: <a.b.c.d>:<port> ** same => n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)}) ** same => n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)}) [Dec 1 12:14:33] -- Executing [postdial@SetVar:1] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutsigip)=192.168.20.226:5065") in new stack [Dec 1 12:14:33] -- Executing [postdial@SetVar:2] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutmediaip)=192.168.20.226:23008") in new stack [Dec 1 12:14:33] -- Executing [postdial@SetVar:3] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutcodec)=g729") in new stack [Dec 1 12:14:33] -- Executing [postdial@SetVar:4] Goto("SIP/office_ServerRoom-00000016", "endsub,1") in new stack [Dec 1 12:14:33] -- Goto (SetVar,endsub,1) [Dec 1 12:14:33] -- Executing [endsub@SetVar:1] Return("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Executing [s@jump2SetVar:2] Return("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1] NoOp("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Auto fallthrough, channel 'SIP/office_ServerRoom-00000016' status is 'UNKNOWN' [Dec 1 12:14:33] -- Remotely bridging SIP/office_Admin2-00000015 and SIP/office_ServerRoom-00000016 When call is terminated the relevant fields in the database for CDR(chanoutsigip), CDR(chanoutmediaip) and CDR(chanoutcodec) are populated with their default values (typically blank or '-----') and NOT with the values above. Am I doing something wrong or is there a different way to populate CDR's with info from called channel (leg B)? Thank you for your replies. Harel -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
