RE: [asterisk-users] Log CODECS in CDR's
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 audioreadformatformat 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 channeltypetechnology 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 [ [EMAIL PROTECTED]:1] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [ [EMAIL PROTECTED]:3] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]: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 [EMAIL PROTECTED]:1] NoOp(SIP/491-0a16d1c0, slin)
in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/491-0a16d1c0, slin)
in new stack
-- Executing [EMAIL PROTECTED]: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.
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RE: [asterisk-users] Log CODECS in CDR's
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729 Dial then negotiates GSM for the outbound
leg,
I want to log both these codecs in a CDR.
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
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dave
cantera
Sent: 11 May 2007 03:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Log CODECS in CDR's
morgan,
I've seen some info on additional variables in the CDR... but haven't
tried it... look to these pages:
daveC
http://www.asterisk.org/doxygen/1.2/AstCDR.html
In addition, you can set your own extra variables by using
Set(CDR(name)=value).
These variables can be output into a text-format CDR by using the
cdr_custom
CDR driver; see the cdr_custom.conf.sample file in the configs directory
for
an example of how to do this.
-and-
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Morgan Gilroy wrote:
Hi,
Does anyone know how to get the codec that was negotiated for a call
after a dial? I want to log them into CDR but can't find any way to do
it without hacking the code.
It would be good if I could get it in an asterisk variable I can log
off
seperatly.
Thanks!
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Re: [asterisk-users] Log CODECS in CDR's
On 5/11/07, Morgan Gilroy [EMAIL PROTECTED] 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 audioreadformatformat 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 channeltypetechnology 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 [EMAIL PROTECTED]:1] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]: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 [EMAIL PROTECTED]:1] NoOp(SIP/491-0a16d1c0, slin) in
new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/491-0a16d1c0, slin) in
new stack
-- Executing [EMAIL PROTECTED]: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.
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RE: [asterisk-users] Log CODECS in CDR's
Hi Morgan,
Am Freitag, den 11.05.2007, 10:32 +0100 schrieb Morgan Gilroy:
Thanks for the pointers, I know about the Set(CDR..) function but I need
the codec that was negotiated in the Dial (once I have that its easy to
stick it into the cdrs as you pointed out).
Ie a call comes in as G729 Dial then negotiates GSM for the outbound
leg,
I want to log both these codecs in a CDR.
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
It is untested, but maybe You can write a little AGI-Script which
accesses some channel vars. Call that AGI as a DeadAGI. A DeadAGI will
be called, if a connection terminates (connect it with the
'h'-Extension, see the wiki). I don't know if the neccessary information
is still alive at this time, but maybe it will do what You want...
HTH,
Karsten
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[asterisk-users] Log CODECS in CDR's
Hi, Does anyone know how to get the codec that was negotiated for a call after a dial? I want to log them into CDR but can't find any way to do it without hacking the code. It would be good if I could get it in an asterisk variable I can log off seperatly. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log CODECS in CDR's
morgan, I've seen some info on additional variables in the CDR... but haven't tried it... look to these pages: daveC http://www.asterisk.org/doxygen/1.2/AstCDR.html In addition, you can set your own extra variables by using Set(CDR(name)=value). These variables can be output into a text-format CDR by using the cdr_custom CDR driver; see the cdr_custom.conf.sample file in the configs directory for an example of how to do this. -and- http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List Morgan Gilroy wrote: Hi, Does anyone know how to get the codec that was negotiated for a call after a dial? I want to log them into CDR but can't find any way to do it without hacking the code. It would be good if I could get it in an asterisk variable I can log off seperatly. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
