[Asterisk-Users] No Sounds

Sat Feb 19 18:14:36 MST 2005

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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