[asterisk-dev] ast_readaudio_callback() called just one time

[Curt Moore]

On Tue, 2007-09-18 at 15:22 -0300, Paulo Garcia wrote:
> What is the difference using Wait or not? Why the vm-login fails to
> play in the first case?

I think what you're experiencing is the lag in the time it takes the RTP
audio to be setup and the streams to begin once the call is established.
The same thing can be accomplished by using:

exten => 4020,1,Answer(500)
exten => 4020,1,VoicemailMain(${CALLERID(num)}@default)

Which will answer the call and start the streams flowing, wait for
500ms, and then continue.  By default, at least for voicemail, Asterisk
does an implicit answer and I think that the channel.c code in trunk was
changed a while back to insert a 500ms wait when a channel was answered
to help with this very problem, ast_answer().  I don't think this change
made it in to 1.4 but I could be wrong.

It's easy enough to insert an Answer(500) in the dialplan for now
whenever there is an application which will answer the channel and play
audio to it to avoid chopping off the first half second or so of audio
if you're experiencing this issue.

There could be something else going on here but this is what I've seen
and how I've worked around it in 1.2 and 1.4.

Cheers,
-Curt
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20070918/395d2cf0/attachment.pgp