[Freeswitch-users] Early Media troubles

[Jock McKechnie]

On Mon, Aug 30, 2010 at 5:48 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> did you try the ones I listed?
>
> I did not phrase them in xml because it's redundant but you should
> simply be able to make a simple extension
> that calls playback WITHOUT calling answer and the file will play in
> early media?
>
> Maybe it's too easy?
>
> This will play a file in early media, answer then play it again as
> in-call media.
>
> <extension name="109">
>  <condition field="destination_number" expression="^109$">
>  <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>
>  <action application="answer"/>
>  <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>
>  </condition>
> </extension>
>
>
I did not, I misunderstood your initial eMail, clearly.

I have rigged up the above and something quite odd happens - definitely much
closer, but still odd. The CLI shows FreeSWITCH accept the call, there is
almost a full ten second pause, and then I hear a 'click' on the line, there
is another pause, and then I get the latter Playback (I've used two
different audio files to ensure I can tell the difference).

However, I've tcpdumped and according to Wireshark:
INVITE
100 Trying
(10.1 second pause)
Early Media RTP
200 OK
Latter Playback RTP
BYE

A review of the audio through WireShark verifies that FS is playing the
first WAV file as, apparently, early media.

So I appear to have two problems: Firstly, there's a ten second pause. I did
not reinstate my sleep statement, so I don't understand why it's waiting so
long after getting the call. And although FreeSWITCH is now definitely
pumping out the early media, Verizon is, apparently, not handing it back. Is
there any possibility something slightly non-standard is happening here, so
VZ's equipment (cough, Sonus, cough) is unaware it should be passing RTP on
to the calling party? I'm afraid my telephony knowledge is primarily VoIP
related to Asterisk (wail, moan, etc, etc) and OpenSIPS. I was under the
impression that a 183 Ringing was required in the flow to alert the far-end
(calling party) that early media is coming, but I could, of course, be very
wrong.

Thank you very much Anthony;

 - Jock
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100831/9dc5a671/attachment-0001.html