[Freeswitch-users] Loss of first second of media

[Anthony Minessale]

its a blue message on cli

It could also be the other side expecting us to send media first or
something silly.
try getting a sip trace and figure out when the rtp starts arriving.


On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
<fraserredmond at gmail.com> wrote:
> 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${destination_number}"/>
>> > <action application="bridge" data="${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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>>
>> AIM: anthm
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>>
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>>
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>
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
MSN:anthony_minessale at hotmail.com
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