asterisk users - Apr 2010 - Full transfer details on inbound calls

Hi

We're using asterisk 1.4.17 using RealTime and my boss has decided that 
we should keep a track of the full history of incoming calls i.e. who 
and when they were transferred to. The asterisk CDR only holds the 
initial answering channel for any call and not any further transfers 
that may have happened.

The idea we are toying with is getting the time and the originating 
channel from the cdr, and then searching the full asterisk logs for the 
channel identifier string. Obviously we would have to have the verbose 
output going to a file and make sure that the verbosity in the console 
is always at least 5.

I've done enough testing to see that is is possible

ish at trinity:/var/log/asterisk$ grep 'Apr 13' full | grep 
SIP/xxx.xxx.xxx.xxx-082090e8 | grep answered
[Apr 13 13:31:11] VERBOSE[17120] logger.c:     -- SIP/811-08214f50 
answered SIP/xxx.xxx.xxx.xxx-082090e8
[Apr 13 13:31:31] VERBOSE[17120] logger.c:     -- SIP/808-08212f08 
answered SIP/xxx.xxx.xxx.xxx-082090e8

The above output shows that the originating channel was answered by sip 
extension 811 and then by 808 20 seconds later.

I am also considering parsing the full log into a mysql database and 
doing the searching in there.

My question is is this a good way to go about what I'm trying to achieve 
or is there a simpler/less process intensive method that I'm missing.

Thanks in advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

Hi,
This may be no use to you if you are using 1.4 but "Call Event
Logging" (or CEL) that is currently in trunk should provide an easier way
to do this.
All events associated with a call e.g. Answer, Hangup, Bridge start, Transfer
etc. are logged to the usual back-ends. We use postgresql via ODBC.
Nic.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Ishfaq Malik
Sent: 13 April 2010 14:50
To: Asterisk
Subject: [asterisk-users] Full transfer details on inbound calls

Hi

We're using asterisk 1.4.17 using RealTime and my boss has decided that 
we should keep a track of the full history of incoming calls i.e. who 
and when they were transferred to. The asterisk CDR only holds the 
initial answering channel for any call and not any further transfers 
that may have happened.

The idea we are toying with is getting the time and the originating 
channel from the cdr, and then searching the full asterisk logs for the 
channel identifier string. Obviously we would have to have the verbose 
output going to a file and make sure that the verbosity in the console 
is always at least 5.

I've done enough testing to see that is is possible

ish at trinity:/var/log/asterisk$ grep 'Apr 13' full | grep 
SIP/xxx.xxx.xxx.xxx-082090e8 | grep answered
[Apr 13 13:31:11] VERBOSE[17120] logger.c:     -- SIP/811-08214f50 
answered SIP/xxx.xxx.xxx.xxx-082090e8
[Apr 13 13:31:31] VERBOSE[17120] logger.c:     -- SIP/808-08212f08 
answered SIP/xxx.xxx.xxx.xxx-082090e8

The above output shows that the originating channel was answered by sip 
extension 811 and then by 808 20 seconds later.

I am also considering parsing the full log into a mysql database and 
doing the searching in there.

My question is is this a good way to go about what I'm trying to achieve 
or is there a simpler/less process intensive method that I'm missing.

Thanks in advance

Ish
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062

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Hi!
> We're using asterisk 1.4.17 using RealTime and my boss has decided that
we
> should keep a track of the full history of incoming calls i.e. who and
> when they were transferred to. The asterisk CDR only holds the initial
> answering channel for any call and not any further transfers that may have
> happened.
> 
> The idea we are toying with is getting the time and the originating 
> channel from the cdr
Depending on the type of phones you use there is also another way to look 
at it: SNOM phones for example have event triggers for certain actions 
that can call a URL, this includes events like off-hook, transfer 
completed and the like.

Taking it even further: You could try trace all SIP signaling to/from 
your Asterisk, or even on your entire LAN, and then try to analyse that. 
I believe there's a number of SIP tools available for that.

Philipp