[asterisk-dev] Problem with 1.4b2 and native sounds

Matt O'Gorman mogorman at digium.com
Tue Sep 26 09:24:01 MST 2006


Hi dan, I tried this again with meetme , and it seemed to work as well.
I could take a look on it on your box but i can't seem to recreate it here.


Mog
----- Original Message -----
From: Dan Austin <Dan_Austin at Phoenix.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Sent: Monday, September 25, 2006 8:31:23 PM GMT-0600
Subject: RE: [asterisk-dev] Problem with 1.4b2 and native sounds


> Hi Dan et all, I tried the conference files without problems, 
> can you try to recreat problem elsewhere or you can contact me 
> directly but it should work.
I suppose it could be a filesystem issue, corruption if you will.
I've moved /var/lib/asterisk/sounds and re-ran make install.  That
did not improve the situation, so I deleted the tar.gz files in
/usr/src/asterisk/sounds and re-ran make sounds; make install
same problem.  I noticed it as the second or more caller into a
meetme.  The 'conf-onlyperson' prompt plays properly, the prompt
"You are caller number" is only noise, followed be a nice clear
number ("2" for the second caller).

I'll have more time to work on it tomorrow, I had family in town 
this weekend, and it's be hard to make time for Asterisk testing....

Dan

> Mog
----- Original Message -----
From: Dan Austin <Dan_Austin at Phoenix.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Sent: Monday, September 25, 2006 11:45:47 AM GMT-0600
Subject: [asterisk-dev] Problem with 1.4b2 and native sounds

I tripped over this last Friday, but only now have I had a chance
to dig into it.

I have been using native format sounds (ulaw, g729) with gsm loaded
as a backup.  Since installing 1.4b2 a number of prompts, mostly related
to conferencing, produce a warbling whitenoise.

To rule out my application  I have been testing direct playback through
the dialplan:

  '8702' =>         1. Answer()                                
                    2. Wait(1)                                 
                    3. Playback(conf-peopleinconf)             
                    4. Hangup()                                

conf-thereare and  conf-peopleinconf show this clearly.  So far all
calls have been using ulaw.  I noticed zttranscode loaded, so I removed
it,
and as likely expected, it made not difference.  I moved the ULAW, G729
and
GSM versions of conf-peopleinconf selectively to force transcoding, no 
improvement.

Removing all versions generates a proper warning that frp app_playback
that the prompt cannot be found.

Is there any other tests I can run to help narrow the issue down?

Dan
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev



More information about the asterisk-dev mailing list