[Asterisk-Users] No Sounds; stumping "The Tryant"

Sun Feb 20 15:00:04 MST 2005

It is weird.. I did a full asterisk reinstall... (no asterisk at home
now)... And well.... Problem persists but this is weird, when it happens, I
reboot the machine, starts working again and sometimes sound stops,
sometimes it doesn’t... This machine seems to have an attitude :)

Last reboot one of the x100p cards complained during the modprobe wcfxo...
:) then it didn’t :) 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 02:07 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Yes, running extra code/libraries/.so means more variables, which we are
trying to eliminate.

Really Anton I am stumped.

Does anyone know? Do you have to have the gsm codec to hear the .gsm sound
files.

Is there an Asterisk at home mailing list?

I found these:
http://www.uninett.no/voip/asterisk.html
No sound on SIP
I had a "allow=all" codecs in the 'sip.conf' while which sort of "stopped"
all sound out. I commented it out, and it was up and running on the sound.
Now I just allow for the g.711 codec with

disallow=all
allow=ulaw 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Sunday, February 20, 2005 2:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

Ok.... Noload modems, alsa and oss... No errors... Is this ok? 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Domingo, 20 de Febrero de 2005 01:18 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

I don’t know if it has something to do but I see 2 mpg123 processes
running:

 3552 pts/1    S      0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3
 3553 pts/1    S      0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 8192
fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3

Everytime I start asterisk..

Also, if I enable alsa I get this error:

 [chan_alsa.so]Feb 20 13:16:44 WARNING[3573]: loader.c:301
__load_resource:
/usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file:
No
such file or directory Feb 20 13:16:44 WARNING[3573]: loader.c:459
load_modules: Loading module chan_alsa.so failed!
[root at voip root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe

And asterisk quits... 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de Febrero de 2005 12:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds; stumping "The Tryant" 

This is getting hard.

So what do we have?

1. The Asterisk server and the phones are using good CODECS.
2. Sound is moving from phone to phone.
3. Sound from the prompts is not playing back to the phones.

So let's go with the first principle: Eliminate the variables.

Do this:

Extensions.conf

[NoSound]

exten => 222,1,Wait(2)
exten => 222,2,Answer
exten => 222,3,Playback(vm-isunavail)
exten => 222,4,Hangup

And see what happens.

I might be missing something. Anyone know how .gsm files are translated to
ulaw/alaw in asterisk?

Race "The Tyrant" Vanderdecken

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 10:38 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Race.

Here are thre results of the tests:

Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x404
(ulaw|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
-
0x0 (nothing) 

Seems both can speak ulaw and ilbc but still, no prompt on the phone :(

Any ideas? 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Correct.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 8:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

This is a very good place to start Race. So if I understand you correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or ulaw/alaw
might help so I can listen to the prompts right?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 06:55 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Grasshopper, 

You have your first clue, the live test works.

Do you understand how SIP works? During the INVITE sequence the Asterisk and
the phone trade RTP CODEC information. RTP is the protocol that actually
carries the sounds, SIP only does the handshaking for the call.
A CODEC is what the RTP is carrying between the pones.

If you do "sip debug" inside of the asterisk command line interface

*CLI> sip debug

Then you will see the SIP Messages and the Codec agreements.

...
16 headers, 13 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT) Found RTP audio format 0 Found RTP
audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP
audio format 101 Peer audio RTP is at port 192.168.1.102:10054 Found
description format pcmu Found description format pcma Found description
format gsm Found description format g729 Found description format
telephone-event
--
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10e
(gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
....
...
...
m=audio 19958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

Asterisk is the " us - 0x8000e (gsm|ulaw|alaw|h263)"
Phone is the " peer - audio=0x10e (gsm|ulaw|alaw|g729)"

Above is the trace of my SNOM 200 -- see the "combined - 0xe
(gsm|ulaw|alaw)"?

The phone can do g729, but asterisk can't, so asterisk and the phone agree
on a non-g729 codec, ulaw. 

Eventually the phone agrees to "a=rtpmap:0 PCMU/8000", it is going to talk
sound using ulaw at 8000hz. (again, I might be a little wrong on the extact
details.)

If the phones agree to use G729 then the playback won't work because you
don't have a g729 license, $10 from Digium.

Remember that asterisk is a third party to a conference and if your
conference is using g729, then asterisk can't do that.

In the sip.conf, 

	Disallow=all
	Allow=gsm
	Allow=ulaw
	Allow=alaw

This will force the phone and asterisk to speak gsm, ulaw or alaw.

I had the same experience with no sound when I first connected a Cisco 7960,
I could here other people, but not the prompts. Asterisk will allow G729 to
pass through, but it will not allow G729 to originate and terminate without
the license (I might be a little mistaken here...)

I hope this helps. I have not use Asterisk at home, it might be different.

Let me know,

Race "The Tyrant" Vanderdecken

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 7:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Hi Race..

In this case, the asterisk|home comes preconfigured with some stuff
different than the asterisk tar file.

I check and the phone supports all mentioned codecs, I also made a test by
using the phone and sjphone to do a live test directly, conversation was
successful using gsm, ulaw and ilbc.

Any other ideas? 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 05:43 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds

Okay, 

A couple of things could be happening so let's run through a list. Your
questions are a little vague so I shall make my answer also vague.

1. Codec. 
	Are you allowing for and does the "phone" support the codec that the
sounds are in? (I.e. do you have a G.729 license for your Minkey?)

2. Does the phone support the G.711a/u law that I think the sounds come in?

3. Test first the voicemail playback demo and make sure you can here the
sound there before jumping into a conference. Dial extension 500, do you
hear anything?

4. Is your phone working? Have you made a call to a live person?

Race

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anton Krall
Sent: Saturday, February 19, 2005 2:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] No Sounds

I just installed asterisk at home to see how it works and seems there is a
problem with sounds... I dont hear any announcements or recordings...
sounds
are on /var/lib/asterisk/sounds and the logs show this:

    -- Created MeetMe conference 1023 for conference '8200'
    -- Playing 'conf-onlyperson' (language 'en')

But I dont hear anything... any experiences with this kind of errors and/or
asterisk at home?

Thx!

__________________________________________________________________
Anton Krall

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