[asterisk-users] Log CODECS in CDR's

[Morgan Gilroy]

That looks like exactly what I want, we are currently on 1.2, ill see if
i can hack similar functionality into it, if not ill have to upgrade to
1.4 (probably best anyway)

 

Thanks for the pointers.

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James
FitzGibbon
Sent: 11 May 2007 15:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Log CODECS in CDR's

 

On 5/11/07, Morgan Gilroy <morgan at telappliant.com> wrote:

	At the moment to find the codecs used I have to look though the
sip
	trace or show channels/show channel (annoying when you have 50+
	channels).
	Im just trying to find an easier and quicker way to keep track
of the 
	codecs used to help with debug etc.
	
	The closest variable iv found is, "${SIP_CODEC} Set the SIP
codec for a
	call"
	Ill see if NoOp (${SIP_CODEC}) shows the codec that was used
without me
	setting it though I don't think it will. 
	
	Iv looked all over and I cant find anything so it looks like I
may have
	to hack a ast_set_var into app_dial or chan_sip



1.4 has the CHANNEL function:

pbxlab-01*CLI> show function CHANNEL 
pbxlab-01*CLI>
  -= Info about function 'CHANNEL' =-

[Syntax]
CHANNEL(item)

[Synopsis]
Gets/sets various pieces of information about the channel.

[Description]
Gets/set various pieces of information about the channel. 
Standard items (provided by all channel technologies) are:
R/O     audioreadformat    format currently being read
R/O     audionativeformat  format used natively for audio
R/O     audiowriteformat   format currently being written 
R/W     callgroup          call groups for call pickup
R/O     channeltype        technology used for channel
R/W     language           language for sounds played
R/W     musicclass         class (from musiconhold.conf ) for hold music
R/W     rxgain             set rxgain level on channel drivers that
support it
R/O     state              state for channel
R/W     tonezone           zone for indications played
R/W     txgain             set txgain level on channel drivers that
support it 
R/O     videonativeformat  format used natively for video

When I put this in a dialplan with NoOps and called channel macros, I
can kind of get what you're describing:

[from-external-pbxtel]
exten   => 491,1,NoOp(${CHANNEL(audioreadformat)}) 
exten   => 491,n,NoOp(${CHANNEL(audiowriteformat)})
exten   => 491,n,NoOp(${CHANNEL(audionativeformat)})
exten   => 491,n,Dial(SIP/491,20,M(logcodec))
exten   => 491,n,Hangup

[macro-logcodec] 
exten => s,1,NoOp(${CHANNEL(audioreadformat)})
exten => s,n,NoOp(${CHANNEL(audiowriteformat)})
exten => s,n,NoOp(${CHANNEL(audionativeformat)})

Console output is:

    -- Executing [ 491 at from-external-pbxtel:1] NoOp("IAX2/pbxtel-01-5",
"ulaw") in new stack
    -- Executing [491 at from-external-pbxtel:2] NoOp("IAX2/pbxtel-01-5",
"ulaw") in new stack
    -- Executing [ 491 at from-external-pbxtel:3] NoOp("IAX2/pbxtel-01-5",
"ulaw") in new stack
    -- Executing [491 at from-external-pbxtel:4] Dial("IAX2/pbxtel-01-5",
"SIP/491|20|M(logcodec)") in new stack 
    -- Called 491
    -- SIP/491-0a16d1c0 is ringing
    -- SIP/491-0a16d1c0 answered IAX2/pbxtel-01-5
    -- Executing [s at macro-logcodec:1] NoOp("SIP/491-0a16d1c0", "slin")
in new stack
    -- Executing [s at macro-logcodec:2] NoOp("SIP/491-0a16d1c0", "slin")
in new stack
    -- Executing [s at macro-logcodec:3] NoOp("SIP/491-0a16d1c0", "gsm") in
new stack
  == Spawn extension (from-external-pbxtel, 491, 4) exited non-zero on
'IAX2/pbxtel-01-5' 
    -- Hungup 'IAX2/pbxtel-01-5'

This is a call coming in as ulaw over IAX2, then going to a SIP
softphone configured for only gsm.

Hope that helps.

-- 
j. 
______________________________________________________________________
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email 
______________________________________________________________________

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070522/f5f01523/attachment.htm