[Asterisk-Users] No Audio on Local Machine, Remote works fine
Hadar Pedhazur
hadar at unorthodox.com
Wed Feb 1 15:09:01 MST 2006
pdhales at optusnet.com.au wrote:
> Is ztdummy loaded properly?
>
> I had a similar problem with a system recently.
The machine has a real Digium T1 card in it, so I didn't think to check
for a timing source. Since it's a "backup" machine, the actual T1 line
isn't plugged in at the moment, but chan_zap.so definitely starts up
correctly.
I'll look into this in the morning (running out of the office now :-).
Thanks for the suggestion!
> PaulH
>
> ----- Original Message -----
> From: "Hadar Pedhazur" <hadar at unorthodox.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Thursday, February 02, 2006 8:16 AM
> Subject: [Asterisk-Users] No Audio on Local Machine, Remote works fine
>
>
>> I don't even know where to begin.
>>
>> I run a lot of production Asterisk servers, for a couple of years now,
>> with no real problems.
>>
>> We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from
>> source tarball(s). Built fine, and started up fine.
>>
>> Any attempts to do local audio (e.g. a "Playback(welcome)") results in
>> complete silence. Worse, the Playback command will hang forever (even if
>> the file is tiny), so it's not just "not being heard", it's like the
>> command is waiting to do something.
>>
>> In one specific case (and only in case), I'll hear a 1/2 second burst of
>> audio, like it's about to start, and then dead air.
>>
>> The "Record" command creates a zero length file if the format is ulaw,
>> and hangs forever after that, and a "wav" format is always 44 bytes
>> before the hang.
>>
>> If I run the "demo-echo-test", I don't hear the prompt, and it hangs on
>> the Playback.
>>
>> OK, now for the weirdness ;-). If I connect this Asterisk to one of our
>> other servers, and dial the echo test on the remote server through this
>> same server, I hear the prompts, and can hear my voice echoed correctly,
>> so this same Asterisk server will happily forward the audio in both
>> directions, it just won't "generate" it. This is with "notransfer=yes",
>> so this Asterisk is staying in the audio stream.
>>
>> I'm stumped, and any help or pointers in the right direction will be
>> greatly appreciated.
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