asterisk users - Apr 2004 - No ringing tone with IAXY (and other bits and bobs)

Hi!

I'm really hope you can help me solve a little mystery, the mystery is
probably just my misunderstanding ! sorry...

I've got an iaxy talking to my * box which connects to two providers.
I'm running the stable release of the pbx.

The only thing is that when dialling from the iaxy the ringing tone isn't
heard while calling someone - you just hear silence then, they either
answer or they don't on the remote end.
>From my extensions.conf is the following - I tried putting the ,r in and
it doesn't help.  Is there some other option I could try here ?

Also I'm getting quite a bit of echo noticed at the remote end as well as
the iaxy end.  All lines are digital, I guess only the jitter buffer is
there to be tweaked to try and help ?

There is also this echo problem with the sipura, but not with an ATA186 or
snom.  The lack of a ringing tone is only with the iaxy.

The Answer,Hangup lines were to solve 'busy' situations with SIP phones,
without this or even with 'Congestion' they just rang forever if a
number
was busy.  They seem to need the 'Answer' line.

If you know a nicer or more correct way for me to do this please let me
know as most times the SIP phone user will hear half a ring and then the
hangup noise generated by the SIP device when a number they call is busy.

Many thanks!!

Chris  

PS please Cc: me a copy as well as to the list in case I miss it -
Thanks.
<< extensions.conf >> 

exten => _00.,1,AbsoluteTimeout(3600)
exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
exten => _00.,3,Answer
exten => _00.,4,Hangup
exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
exten => _00.,104,Answer
exten => _00.,105,Hangup

<<iax.conf>>

[iaxy]
type=friend
accountcode=iaxy
disallow=all
;;allow=adpcm
allow=ulaw
username=iaxy
secret=xxx
auth=md5
nat=yes         <- nat=1 ??
notransfer=yes  <-this doesn't seem to work, perhaps in the wrong order?
host=dynamic
qualify=10000

Is the definitive order these should be in listed anywhere as I know it
really seems critical and lines can be ignored if they're not in spot on
the right order?

What version of the Asterisk code are you running? 1_0 stable is definitely
broken wrt ringback, and the latest stuff seems really broken in all kinds
of ways. After seeing that others were having similar problems, and that
someone had solved many of them by rolling back to the CVS version from 3/5,
I tried the same and things are working marvelously (well, mostly).

-brian 
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com 
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Chris Orme
> Sent: Saturday, April 10, 2004 6:37 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] No ringing tone with IAXY (and 
> other bits and bobs)
> 
> Hi!
> 
> I'm really hope you can help me solve a little mystery, the 
> mystery is probably just my misunderstanding ! sorry...
> 
> I've got an iaxy talking to my * box which connects to two providers.
> I'm running the stable release of the pbx.
> 
> The only thing is that when dialling from the iaxy the 
> ringing tone isn't heard while calling someone - you just 
> hear silence then, they either answer or they don't on the remote end.
> 
> >From my extensions.conf is the following - I tried putting the ,r in 
> >and
> it doesn't help.  Is there some other option I could try here ?
> 
> Also I'm getting quite a bit of echo noticed at the remote 
> end as well as the iaxy end.  All lines are digital, I guess 
> only the jitter buffer is there to be tweaked to try and help ?
> 
> There is also this echo problem with the sipura, but not with 
> an ATA186 or snom.  The lack of a ringing tone is only with the iaxy.
> 
> The Answer,Hangup lines were to solve 'busy' situations with 
> SIP phones, without this or even with 'Congestion' they just 
> rang forever if a number was busy.  They seem to need the 
> 'Answer' line.
> 
> If you know a nicer or more correct way for me to do this 
> please let me know as most times the SIP phone user will hear 
> half a ring and then the hangup noise generated by the SIP 
> device when a number they call is busy.
> 
> Many thanks!!
> 
> Chris  
> 
> PS please Cc: me a copy as well as to the list in case I miss 
> it - Thanks.
> << extensions.conf >> 
> 
> exten => _00.,1,AbsoluteTimeout(3600)
> exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
> exten => _00.,3,Answer
> exten => _00.,4,Hangup
> exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
> exten => _00.,104,Answer
> exten => _00.,105,Hangup
> 
> <<iax.conf>>
> 
> [iaxy]
> type=friend
> accountcode=iaxy
> disallow=all
> ;;allow=adpcm
> allow=ulaw
> username=iaxy
> secret=xxx
> auth=md5
> nat=yes         <- nat=1 ??
> notransfer=yes  <-this doesn't seem to work, perhaps in the 
> wrong order?
> host=dynamic
> qualify=10000
> 
> Is the definitive order these should be in listed anywhere as 
> I know it really seems critical and lines can be ignored if 
> they're not in spot on the right order?
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi all,
I am not sure if tis is a bug but..
Was learning about VM etc to see how it all worked, and I noticed the
following..

In the default install, the VM system leaves 3 different copies of the Voice
message.
Size	filename
13kb	Msg0000.gsm
13kb	Msg0000.wav
122kb	Msg0000.WAV  <- under UNIX we have case sensitive file names of
course.

I wanted to have a look at these files so loaded them into SOUND FORGE 6.
This first thing I noticed was that the LARGER file is of much HIGHER
volume. Like it had been normalised to 100%
The smaller was file, when loaded into sound forge, did not load properly,
only the first 2 seconds loads.

Can anyone explain these issues and why they exist?

All in all, I was wondering what would be the best format for best quality
but with still great compression.

I want to archive all calls for a period of time with self expire. (For
example dedicate 5 gig disk space to the last number of calls that can fit
in the 5gig.) I want to store the best quality possible but also make best
use of disk space, so I can store for even longer periods.  I was
considering ogg but then is occurred to me that GSM or other codecs designed
for audio with this frequency response may be better. (But the GSM file
above is not as clear as the WAV ones produced.)

I was also wondering if the VM system when emailing the audio can be setup
to use something like ogg or MP3?

Comments appreciated,
James Gardiner

On Tuesday 13 April 2004 12:12 am, James Gardiner wrote:
> Hi all,
> I am not sure if tis is a bug but..
> Was learning about VM etc to see how it all worked, and I noticed the
> following..
>
> In the default install, the VM system leaves 3 different copies of the
> Voice message.
> Size	filename
> 13kb	Msg0000.gsm
> 13kb	Msg0000.wav
> 122kb	Msg0000.WAV  <- under UNIX we have case sensitive file names of
> course.
>
> I wanted to have a look at these files so loaded them into SOUND FORGE 6.
> This first thing I noticed was that the LARGER file is of much HIGHER
> volume. Like it had been normalised to 100%
> The smaller was file, when loaded into sound forge, did not load properly,
> only the first 2 seconds loads.
>
> Can anyone explain these issues and why they exist?
>
> All in all, I was wondering what would be the best format for best quality
> but with still great compression.
>
> I want to archive all calls for a period of time with self expire. (For
> example dedicate 5 gig disk space to the last number of calls that can fit
> in the 5gig.) I want to store the best quality possible but also make best
> use of disk space, so I can store for even longer periods.  I was
> considering ogg but then is occurred to me that GSM or other codecs
> designed for audio with this frequency response may be better. (But the GSM
> file above is not as clear as the WAV ones produced.)
>
> I was also wondering if the VM system when emailing the audio can be setup
> to use something like ogg or MP3?
>
> Comments appreciated,
> James Gardiner
That is a very interesting observation I had not seen yet.

There are certainly a few ways to do this.  Here is an idea: run a cron job that
downsamples the WAV file using sox, then compresses it, and deletes the
redundant
files.  Make sure to have the cron job run during the least-busy part of the
day.

Yes, it is a crude solution.  ;)  Yet it may get you by until you get a better
solution.

Anon

All in all, I was more hoping to get some words of wisdom from the more
worldly Audio Compression experienced people in regard of the question below
about what is the best way to store audio recorded with asterisk.

Ie, to keep the BEST possible quality asterisk can record but still getting
great compression.  And not having to use any real time compression formats.
(mp3, ogg, etc)

Thanks,
James
 
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com 
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Anon
> Sent: Thursday, 15 April 2004 11:15 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] VoiceMailBox wav file format in EMAIL.
> 
> On Tuesday 13 April 2004 12:12 am, James Gardiner wrote:
> > Hi all,
> > I am not sure if tis is a bug but..
> > Was learning about VM etc to see how it all worked, and I 
> noticed the 
> > following..
> >
> > In the default install, the VM system leaves 3 different 
> copies of the 
> > Voice message.
> > Size	filename
> > 13kb	Msg0000.gsm
> > 13kb	Msg0000.wav
> > 122kb	Msg0000.WAV  <- under UNIX we have case 
> sensitive file names of
> > course.
> >
> > I wanted to have a look at these files so loaded them into 
> SOUND FORGE 6.
> > This first thing I noticed was that the LARGER file is of 
> much HIGHER 
> > volume. Like it had been normalised to 100% The smaller was 
> file, when 
> > loaded into sound forge, did not load properly, only the first 2 
> > seconds loads.
> >
> > Can anyone explain these issues and why they exist?
> >
> > All in all, I was wondering what would be the best format for best 
> > quality but with still great compression.
> >
> > I want to archive all calls for a period of time with self expire. 
> > (For example dedicate 5 gig disk space to the last number of calls 
> > that can fit in the 5gig.) I want to store the best quality 
> possible 
> > but also make best use of disk space, so I can store for 
> even longer 
> > periods.  I was considering ogg but then is occurred to me 
> that GSM or 
> > other codecs designed for audio with this frequency response may be 
> > better. (But the GSM file above is not as clear as the WAV ones 
> > produced.)
> >
> > I was also wondering if the VM system when emailing the 
> audio can be 
> > setup to use something like ogg or MP3?
> >
> > Comments appreciated,
> > James Gardiner
> That is a very interesting observation I had not seen yet.
> 
> There are certainly a few ways to do this.  Here is an idea: 
> run a cron job that downsamples the WAV file using sox, then 
> compresses it, and deletes the redundant files.  Make sure to 
> have the cron job run during the least-busy part of the day.
> 
> Yes, it is a crude solution.  ;)  Yet it may get you by until 
> you get a better solution.
> 
> Anon
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>