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Summary:ASTERISK-01653: ALSA Device or resource busy error with CVS HEAD
Reporter:djflux (djflux)Labels:
Date Opened:2004-05-19 12:01:07Date Closed:2011-06-07 14:04:39
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Related
Issues:
Environment:Attachments:
Description:When trying to call from an H.323 hardphone through an H.323 trunk group directed to Asterisk from a Definity Prologix to a SIP softphone, Asterisk gives the error in the Additional Information field of this bug.  Below is a visual representation of the setup

Hardphone --> Definity --> Asterisk --> SIP Client

There are no other processes using those devices (/dev/snd/pcm*) according to an lsof.  The following is the call process:

1. Dial extension 1609 from the hardphone
2. Prologix passses any call to 1609 down an H.323 trunk group to the Asterisk server
3. SIP softphone rings
4. Answer SIP client and connection immediately is disconnected and error is displayed on the Asterisk console.

Here are the details of my configuration and equipment:

Avaya 4612IP Hardphone
Avaya Definity Prologix R12 with MedPro and CLAN
SIP clients: Windows Messenger 4.7.2009, X-Lite 1103a
Asterisk CVS-HEAD-5/10/04-20:43:43 and CVS-HEAD-5/19/04-10:18:12
OpenH323 1.13.5 Janus patch 2 compiled from source
PWLib 1.6.6 Janus patch 2 compiled from source
Fedora Core 1 with updates
kernel-2.4.22-1.2188.nptl_48.rhfc1.at
kernel-module-alsa-2.4.22-1.2188.nptl_48.rhfc1.at-1.0.4-23.rhfc1.at
alsa-driver-1.0.4-23.rhfc1.at
alsa-lib-1.0.4-12.rhfc1.at
alsa-utils-1.0.4-7.rhfc1.at

This problem does not occurr when using openh323-1.12.2/pwlib-1.5.2 with the same CVS versions of Asterisk.  The call completes successfully when using the above version of openh323 and pwlib (there is only one-way audio, but that is another issue).


****** ADDITIONAL INFORMATION ******

Error
---------------------------------------------
ALSA lib pcm_hw.c:1057:(snd_pcm_hw_open) open /dev/snd/pcmC0D0c failed: Device or resource busy
 3:11.518          H225 Answer:8856f20       h323ep.cxx(2143)  Codec   Could not open sound channel "Ensoniq AudioPCI" for recording:
 3:11.518          H225 Answer:8856f20     channels.cxx(1096)  LogChan Transmit thread aborted (open fail) for G.711-uLaw-64k{sw} <1>

-----------------------------
sip.conf
-----------------------------
                                                                                                                                                     
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
disallow=all
allow=ulaw
canreinvite=no
                                                                                                                                                     
[1609]
type=friend
host=dynamic
username=1609
secret=password
mailbox=1609
canreinvite=no
nat=no
                                                                                                                                                     
                                                                                                                                                     
-----------------------------
h323.conf
-----------------------------
[general]
port = 1720
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=inband
context=default
Comments:By: jerjer (jerjer) 2004-05-19 16:57:22

we need more debug and specifically what H.323 channel driver are you running?
By: djflux (djflux) 2004-05-19 17:00:17

My apologies.  I'm using chan_h323.so that comes in the CVS of Asterisk.  What trace level of h.323 should I set to give you the debug information that you need?
By: Mark Spencer (markster) 2004-05-19 17:07:13

Excuse me for commenting on H.323 (I *never* like to be involved with H.323), but let me make some observations:

Obviously the h323 library is trying to open the sound device directly, this isn't even something from within asterisk.  Given that Asterisk is working with the non-patched version of H.323, may I ask what suggests to you that this is an Asterisk bug?
By: jerjer (jerjer) 2004-05-19 17:07:33

read the readme included in the /path/to/asterisk/channels/H323 directory
By: jerjer (jerjer) 2004-05-19 17:09:29

not a bug. RTFM
By: djflux (djflux) 2004-05-19 17:12:00

I've read the README in the h323 directory.  I assumed that since in the README it states that the "code runs on Open H.323 and PWLib Janus (Patch 2)," that the bug lies with Asterisk.  I am running the versions listed in the README.
By: djflux (djflux) 2004-05-19 17:15:56

I would like a more descriptive reason as to why this is not a bug if I am running the versions of code listed in the README in channels/h323.  Thank you
By: djflux (djflux) 2004-05-19 17:18:46

The reason that I wanted to try a different version of OpenH323 and PWLib is because in the older versions (1.12.2 and 1.5.2) I am only getting one-way audio and figured that problem may lie in Asterisk being compiled with older libraries.
By: jerjer (jerjer) 2004-05-19 17:29:47

cvs update
By: Matthew Fredrickson (mattf) 2004-05-19 17:31:20

Basically, what Mark is saying is that this is not an ASTERISK bug; it is a bug in OpenH323.  

You should probably report this to the maintainers of the OpenH323 library.

AGAIN, if you are not running the recommended versions of OpenH323 or PWLib you're best make a feature request to the chan_h323 maintainer about this.
By: djflux (djflux) 2004-05-19 17:40:53

JerJer ... which CVS should I update?  Asterisk?  OpenH323/PWLib?
By: djflux (djflux) 2004-05-19 17:48:05

mattf, thanks for the clarification.  I just did an update on the README in h323 and JerJer changed it so that it only reads 1.12.2 and 1.5.2 as supported.  I'll check with the OpenH323 guys.

JerJer, sorry to be a pain, but I always read documentation and do research before opening bugs so it was a little discouraging when the bug was closed with RTFM.  

Thanks guys.
By: jerjer (jerjer) 2004-05-19 17:51:40

Resolved is for acutal bugs