[Asterisk-Users] No sound into asterisk???

[Race Vanderdecken]

Greetings,

	Try a couple of basic things here.

	CLI> sip debug

	This will let you see if SIP and RTP are really talking. If you
need help with the SIP messages let me know


	Often there is a CODEC miss match. Asterisk prompts are in GSM,
what is your phone producing?

	Look for SIP messages 403 and 488, these are CODEC mismatches (A
codec is the compression used to digitize the analog sounds.)

	Remember to run asterisk

	[user]# asterisk  -dvvvvvvgc  

	you have to put the 'd' first to get debug to work every time.


	Race Vanderdecken
	
	Asterisk  \at Vanderdecken  \dot  com

	

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Noah
Miller
Sent: Friday, September 24, 2004 10:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] No sound into asterisk???

Hi -

I think I might have seen this problem on the list before, so I'm sorry 
if this is a duplicate, but I couldn't find it when searching through 
the archive....

I'm just setting up a new machine with asterisk.  It's a RH9 box, and 
I've tried the RC2 tarball, the 1.0 CVS and the 1.0 RPM's from nacs.net 
(thanks).  My config is basically the sample barebones sip setup from 
the O'Reilly site (onlamp.com).  The exact problem is that I can get 
sound out of asterisk to my sip extensions, but asterisk is not able to 
get any sound from the sip devices.  The console error I get is:

Sep 24 09:33:03 WARNING[1111635520]: app.c:599 ast_play_and_record: No 
audio available on SIP/04-e2f4??
     -- User hung up

I've only tried soft phones so far, but I have tried several different 
ones, and all of them seem to have the same problem.  I'm pretty sure 
the problem is with my * config, or some missing libraries on my 
server.  Other than that, I'm clueless.

Any ideas?

Thanks!
Noah

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