Re: [asterisk-users] CHANNEL arguments documentation?

2012-08-25 Thread Stefan at WPF
Looking at an older Channel documentation, it seems like the "all"
parameter wasn't adapted to the new, additional parameters. Will fill a bug
report during the day.

2012/8/25 Stefan at WPF 

> Thank you richard, now I understand and it works, except that all only
> gives
>
>> ssrc
>>
>> themssrc
>>
>> lp
>>
>> rxjitter
>>
>> rxcount
>>
>> txjitter
>>
>> txcount
>>
>> rlp
>>
>> rtt
>>
>> and not all mentioned fields.
>
> Do you know if there is a updated documentation anywhere?
> I can guess what most of the fields are, but I am not an expert on these
> things ;-)
> I especially have problems with the remote_x and local_x fields: if it's
> prefixed with local, it means send from me to a client? or does it mean
> things that didn't arrive locally, therefore send from client to me?
> tx should be always from me to client and rx the other way round, right?
>
>
> 2012/8/24 Richard Mudgett 
>
>> > Using
>> >
>> >
>> > exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
>> >
>> > gives me
>> >
>> >
>> > [Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
>> > sip_acf_channel_read: Unrecognized argument
>> > 'rtpqos,audio,local_lostpackets' to CHANNEL
>> >
>> > [Aug 24 12:08:10] WARNING[12087]: func_channel.c:393
>> > func_channel_read: Unknown or unavailable item requested:
>> > 'rtpqos,audio,local_lostpackets'
>> >
>> > According to
>> >
>> http://lists.digium.com/pipermail/asterisk-users/2011-September/266075.html
>> > the variable names have changed, however I couldn't find an updated
>> > documentation or understand from the mail reply, what I have to
>> > change in the example above.
>> > So, can somebody please tell me where to get an updated documentation
>> > or how to fix my problem? I am using Asterisk 1.8.10.1.
>> >
>>
>> This is what the chan_sip code will accept in the third parameter
>> position:
>>
>> "txcount",
>> "rxcount",
>> "txjitter",
>> "rxjitter",
>> "remote_maxjitter",
>> "remote_minjitter",
>> "remote_normdevjitter",
>> "remote_stdevjitter",
>> "local_maxjitter",
>> "local_minjitter",
>> "local_normdevjitter",
>> "local_stdevjitter",
>> "txploss",
>> "rxploss",
>> "remote_maxrxploss",
>> "remote_minrxploss",
>> "remote_normdevrxploss",
>> "remote_stdevrxploss",
>> "local_maxrxploss",
>> "local_minrxploss",
>> "local_normdevrxploss",
>> "local_stdevrxploss",
>> "rtt",
>> "maxrtt",
>> "minrtt",
>> "normdevrtt",
>> "stdevrtt",
>> "local_ssrc",
>> "remote_ssrc",
>>
>> It also will accept nothing or "all".
>>
>> Richard
>>
>> --
>> _
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] CHANNEL arguments documentation?

2012-08-25 Thread Stefan at WPF
Thank you richard, now I understand and it works, except that all only gives

> ssrc
>
> themssrc
>
> lp
>
> rxjitter
>
> rxcount
>
> txjitter
>
> txcount
>
> rlp
>
> rtt
>
> and not all mentioned fields.

Do you know if there is a updated documentation anywhere?
I can guess what most of the fields are, but I am not an expert on these
things ;-)
I especially have problems with the remote_x and local_x fields: if it's
prefixed with local, it means send from me to a client? or does it mean
things that didn't arrive locally, therefore send from client to me?
tx should be always from me to client and rx the other way round, right?


2012/8/24 Richard Mudgett 

> > Using
> >
> >
> > exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
> >
> > gives me
> >
> >
> > [Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
> > sip_acf_channel_read: Unrecognized argument
> > 'rtpqos,audio,local_lostpackets' to CHANNEL
> >
> > [Aug 24 12:08:10] WARNING[12087]: func_channel.c:393
> > func_channel_read: Unknown or unavailable item requested:
> > 'rtpqos,audio,local_lostpackets'
> >
> > According to
> >
> http://lists.digium.com/pipermail/asterisk-users/2011-September/266075.html
> > the variable names have changed, however I couldn't find an updated
> > documentation or understand from the mail reply, what I have to
> > change in the example above.
> > So, can somebody please tell me where to get an updated documentation
> > or how to fix my problem? I am using Asterisk 1.8.10.1.
> >
>
> This is what the chan_sip code will accept in the third parameter position:
>
> "txcount",
> "rxcount",
> "txjitter",
> "rxjitter",
> "remote_maxjitter",
> "remote_minjitter",
> "remote_normdevjitter",
> "remote_stdevjitter",
> "local_maxjitter",
> "local_minjitter",
> "local_normdevjitter",
> "local_stdevjitter",
> "txploss",
> "rxploss",
> "remote_maxrxploss",
> "remote_minrxploss",
> "remote_normdevrxploss",
> "remote_stdevrxploss",
> "local_maxrxploss",
> "local_minrxploss",
> "local_normdevrxploss",
> "local_stdevrxploss",
> "rtt",
> "maxrtt",
> "minrtt",
> "normdevrtt",
> "stdevrtt",
> "local_ssrc",
> "remote_ssrc",
>
> It also will accept nothing or "all".
>
> Richard
>
> --
> _
> -- 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
>
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Re: [asterisk-users] CHANNEL arguments documentation?

2012-08-24 Thread Richard Mudgett
> Using
> 
> 
> exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
> 
> gives me
> 
> 
> [Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221
> sip_acf_channel_read: Unrecognized argument
> 'rtpqos,audio,local_lostpackets' to CHANNEL
> 
> [Aug 24 12:08:10] WARNING[12087]: func_channel.c:393
> func_channel_read: Unknown or unavailable item requested:
> 'rtpqos,audio,local_lostpackets'
> 
> According to
> http://lists.digium.com/pipermail/asterisk-users/2011-September/266075.html
> the variable names have changed, however I couldn't find an updated
> documentation or understand from the mail reply, what I have to
> change in the example above.
> So, can somebody please tell me where to get an updated documentation
> or how to fix my problem? I am using Asterisk 1.8.10.1.
> 

This is what the chan_sip code will accept in the third parameter position:

"txcount",   
"rxcount",   
"txjitter",  
"rxjitter",  
"remote_maxjitter",  
"remote_minjitter",  
"remote_normdevjitter",  
"remote_stdevjitter",
"local_maxjitter",   
"local_minjitter",   
"local_normdevjitter",   
"local_stdevjitter", 
"txploss",   
"rxploss",   
"remote_maxrxploss", 
"remote_minrxploss", 
"remote_normdevrxploss", 
"remote_stdevrxploss",   
"local_maxrxploss",  
"local_minrxploss",  
"local_normdevrxploss",  
"local_stdevrxploss",
"rtt",   
"maxrtt",
"minrtt",
"normdevrtt",
"stdevrtt",  
"local_ssrc",
"remote_ssrc",   

It also will accept nothing or "all".

Richard

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Re: [asterisk-users] CHANNEL arguments documentation?

2012-08-24 Thread Stefan at WPF
Thank you Eric, but this is still the old documentation:
>
>  rtpqos - R/O Get QOS information about the RTP stream
>
> This option takes two additional arguments:
>
> Argument 1:
>
>  'audio' Get data about the audio stream
>
>  'video' Get data about the video stream
>
>  'text'  Get data about the text stream
>
> Argument 2:
>
>  'local_ssrc'Local SSRC (stream ID)
>
>  'local_lostpackets' Local lost packets
>
> According to this,

> CHANNEL(rtpqos,audio,local_lostpackets)

would be correct

>



2012/8/24 Eric Wieling 

> pbx*CLI> core show function CHANNEL
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
> Sent: Friday, August 24, 2012 6:37 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] CHANNEL arguments documentation?
>
> Using
>
>
> exten =>
> h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)})
>
> gives me
>
>
> [Aug 24 12:08:10] WARNING[12087]:
> sip/dialplan_functions.c:221 sip_acf_channel_read: Unrecognized argument
> 'rtpqos,audio,local_lostpackets' to CHANNEL
>
> [Aug 24 12:08:10] WARNING[12087]: func_channel.c:393
> func_channel_read: Unknown or unavailable item requested:
> 'rtpqos,audio,local_lostpackets'
>
> According to
> http://lists.digium.com/pipermail/asterisk-users/2011-September/266075.htmlthe
>  variable names have changed, however I couldn't find an updated
> documentation or understand from the mail reply, what I have to change in
> the example above.
> So, can somebody please tell me where to get an updated documentation or
> how to fix my problem? I am using Asterisk 1.8.10.1.
>
> Thanks :-)
>
> --
> _
> -- 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
>
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Re: [asterisk-users] CHANNEL arguments documentation?

2012-08-24 Thread Eric Wieling
pbx*CLI> core show function CHANNEL

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan at WPF
Sent: Friday, August 24, 2012 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] CHANNEL arguments documentation?

Using


exten => h,n,set(CDR(llp)=${CHANNEL(rtpqos,audio,local_lostpackets)}) 

gives me 
 

[Aug 24 12:08:10] WARNING[12087]: sip/dialplan_functions.c:221 
sip_acf_channel_read: Unrecognized argument 'rtpqos,audio,local_lostpackets' to 
CHANNEL

[Aug 24 12:08:10] WARNING[12087]: func_channel.c:393 
func_channel_read: Unknown or unavailable item requested: 
'rtpqos,audio,local_lostpackets'

According to 
http://lists.digium.com/pipermail/asterisk-users/2011-September/266075.html the 
variable names have changed, however I couldn't find an updated documentation 
or understand from the mail reply, what I have to change in the example above.
So, can somebody please tell me where to get an updated documentation or how to 
fix my problem? I am using Asterisk 1.8.10.1.

Thanks :-) 

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