[Asterisk-Users] First second choppy

[Robert Goodyear]

>
>>
>> On Mar 28, 2005, at 3:22 PM, Noah Silverman wrote:
>>
>>> Hi,
>>>
>>> When someone calls into our * system over a PTSN line, we answer 
>>> with a recorded prompt.  (Thank  you for calling, etc..)
>>>
>>> The first second of this prompt ALWAYS skips.  After that, 
>>> everything sounds great and works perfectly.  There is nothing wrong 
>>> with the prompt.
>>
>>
>> Yeah, there's some wackyness (wackiness?) during call setup, and it 
>> usually manifests itself as a big ugly dropout on an outbound call 
>> too. I have been experimenting with the whole RINGING, WAIT, ANSWER 
>> sequence of steps at the beginning of my inbound contexts to get just 
>> the right amount of time to settle the call down.
>>
>> However, I'm not sure if I should be doing RINGING, ANSWER, WAIT or 
>> some other sequence of events, or just RINGING(n) then ANSWER.
>>
>> Anyone know if WAIT is not advisable to workaround the problem Noah's 
>> asking about?
>>
>> /rg
> Thanks Rob.  Let me know if  you come up with anything.
>
> Another option would be to ANSWER and then play one second of silence. 
>  If there is "chop" during that second, nobody will notice.
>
> -N

Agreed, but what I'd really like to to in my case is to present ringing 
tone *longer* and then get right into the greeting. Same goes with 
picking up too quickly; would my "Hello, this is Rob." come out as 
"...ob." if I wired some more DIDs straight to my SIP phones? That's 
what I'd like to work around myself while reducing (apparent) latency 
to the caller.