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 at 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 at 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(${CHANNEL(peername)},port)})
** same => n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same => n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec 1 12:14:33] -- Executing [postdial at SetVar:1]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutsigip)=192.168.20.226:5065") in new stack
[Dec 1 12:14:33] -- Executing [postdial at SetVar:2]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutmediaip)=192.168.20.226:23008") in new stack
[Dec 1 12:14:33] -- Executing [postdial at SetVar:3]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutcodec)=g729") in new stack
[Dec 1 12:14:33] -- Executing [postdial at 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 at SetVar:1]
Return("SIP/office_ServerRoom-00000016", "") in new stack
[Dec 1 12:14:33] -- Executing [s at jump2SetVar:2]
Return("SIP/office_ServerRoom-00000016", "") in new stack
[Dec 1 12:14:33] -- Executing [s at 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
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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 at 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 at 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(peername)},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 at SetVar:1]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutsigip)=192.168.20.226:5065") in new stack
[Dec 1 12:14:33] -- Executing [postdial at SetVar:2]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutmediaip)=192.168.20.226:23008") in new stack
[Dec 1 12:14:33] -- Executing [postdial at SetVar:3]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutcodec)=g729") in new stack
[Dec 1 12:14:33] -- Executing [postdial at 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 at SetVar:1]
Return("SIP/office_ServerRoom-00000016", "") in new stack
[Dec 1 12:14:33] -- Executing [s at jump2SetVar:2]
Return("SIP/office_ServerRoom-00000016", "") in new stack
[Dec 1 12:14:33] -- Executing [s at 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
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Le 06/12/2011 10:16, Harel Cohen a ?crit :> > Hello Everyone, >Hi Harel> 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? >Please take a look at issue ASTERISK-18875 <https://issues.asterisk.org/jira/browse/ASTERISK-18875> [...] -- Daniel
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" <danny at debsinc.com>
Subject: Re: [asterisk-users] Populate CDR issues
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <009801ccb41b$24417cd0$6cc47670$@debsinc.com>
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Harel Cohen
Sent: Tuesday, December 06, 2011 3:16 AM
To: asterisk-users at lists.digium.com
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 at 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 at 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 at SetVar:1]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutsigip)=192.168.20.226:5065") in new stack
[Dec 1 12:14:33] -- Executing [postdial at SetVar:2]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutmediaip)=192.168.20.226:23008") in new stack
[Dec 1 12:14:33] -- Executing [postdial at SetVar:3]
Set("SIP/office_ServerRoom-00000016",
"CDR(chanoutcodec)=g729") in new stack
[Dec 1 12:14:33] -- Executing [postdial at 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 at SetVar:1]
Return("SIP/office_ServerRoom-00000016", "") in new stack
[Dec 1 12:14:33] -- Executing [s at jump2SetVar:2]
Return("SIP/office_ServerRoom-00000016", "") in new stack
[Dec 1 12:14:33] -- Executing [s at 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
Hi Mike, I've tried updating my CDR's via the h exten but with no success. I've tried with both endbeforehexten=no and endbeforehexten=yes (in cdr.conf) but the value refused to appear in my CDR (even though I see the Set() application being executed in the console under the h exten). Thank you for your suggestion though... Any other thoughts are welcome. Kind Regards, Harel Cohen -----Original Message----- Date: Mon, 12 Dec 2011 13:41:31 -0700 From: Mike Diehl <mdiehl at diehlnet.com> Subject: Re: [asterisk-users] Populate CDR issues To: asterisk-users at lists.digium.com Cc: Harel Cohen <harel at easycall.gi> Message-ID: <201112121341.32142.mdiehl at diehlnet.com> Content-Type: Text/Plain; charset="iso-8859-1" On Monday 12 December 2011 4:28:17 am Harel Cohen wrote:> 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.I agree, this is a perfectly valid use of the CDR. I do the same thing, btw. I think what you are seeing is that when your call starts, Asterisk creates a record, either in memory, or in a db transaction. When the call is torn down, the record is updated and committed to the db. The down-shot is that any changes you make to the db record get clobbered by this last update. I ended up making some of my updates in the hang-up phase via the "h" extension. See if that will do what you need. -- Take care and have fun, Mike Diehl. ------------------------------ Date: Thu, 1 Dec 2011 12:57:56 +0100 From: Harel Cohen <harel at easycall.gi> Subject: [asterisk-users] Populate CDR issues To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Message-ID: <CABC006234837141A279831F68041406CABE9415A4 at ECS.EasyCall.local> Content-Type: text/plain; charset="us-ascii" 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 at 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 at 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(peername)},port)}) ** same => n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)}) ** same => n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)}) [Dec 1 12:14:33] -- Executing [postdial at SetVar:1] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutsigip)=192.168.20.226:5065") in new stack [Dec 1 12:14:33] -- Executing [postdial at SetVar:2] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutmediaip)=192.168.20.226:23008") in new stack [Dec 1 12:14:33] -- Executing [postdial at SetVar:3] Set("SIP/office_ServerRoom-00000016", "CDR(chanoutcodec)=g729") in new stack [Dec 1 12:14:33] -- Executing [postdial at 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 at SetVar:1] Return("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Executing [s at jump2SetVar:2] Return("SIP/office_ServerRoom-00000016", "") in new stack [Dec 1 12:14:33] -- Executing [s at 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
Hi, http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting Read it, Might be it wil solved your doubts. On Fri, Dec 30, 2011 at 12:01 AM, Matt Hamilton <mistral9999 at hotmail.com>wrote:> I have a couple of performance/memory related questions: > > > Is there any downside to using long URIs as far as memory or database > (mysql) performance is concerned, e.g. > > sip:1234567890_1234567890 at abc.com? Or is this negligible? > > > Also is there a performance hit if no pattern matching is used? e.g. > > exten => _XXX,Noop(... > > vs > > exten => 100,Noop(.. > exten => 101,Noop(... > exten => 102,Noop(... > ... > exten => 999,Noop(... > > If a call comes to 999, does Asterisk go through each extension > sequentially from 100 to 999 until it finds the matching one? > > Thanks, > Matt > > > > > -- > _____________________________________________________________________ > -- 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 >-- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120102/48b2de18/attachment.htm>