[Freeswitch-users] Loss of first second of media

[Fraser Redmond]

Sorry, yes, I am setting ignore_early_media=true in the first area. (Or are
you saying that should be off? I forget now why I needed it on, but there
was a reason I added it.)

Yes, the bridge doesn't start until after the A-leg has answered.

Thanks for the suggestion about nat/auto-changing port, I'll have a look
into that - would that be in the cli output or in a sip trace? I've already
looked and it's not appearing in the CLI output (with loglevel=debug),
haven't looked in the sip trace yet.

Cheers,
Fraser




On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> are you setting ignore_early_media=true in the first vars=values area?
>
> This looks like you could be calling one leg who is still not answered
> and then bridging it to another dest.  The bridge app will wait for
> the first leg to answer before bridging.
>
> Also if you have any NAT anywhere, look for an "auto-changing port"
> type message which can also be attributed to this due to a detection
> period for incorrect ports.
>
>
>
> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
> > event_socket:
> > api originate {vars=values}user/$fromExtn at Domain
> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
> >
> > then
> >
> > dialplan:
> > <action application="set" data="effective_caller_id_number=+1800number"/>
> > <action application="set" data="effective_caller_id_name="/>
> > (set and/or export a bunch of other vars too)
> > <action application="set"
> > data="dial_string=sofia/gateway/gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> > <action application="bridge" data="${dial_string}"/>
> >
> > Cheers,
> > Fraser
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> how are you accomplishing that? by which technique?
> >>
> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >> <fraserredmond at gmail.com> wrote:
> >> > The call is originated from Freeswitch (via CLI) to a softphone, then
> >> > when
> >> > that is connected it bridges out to the gateway.
> >> >
> >> > Cheers,
> >> > Fraser
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> Where is the other side of this call coming from?
> >> >>
> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >> >>
> >> >> What goes in the empty space above?
> >
> >
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>
>
>
> --
> Anthony Minessale II
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