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
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@gmail.com> wrote:
> event_socket:
> api originate {vars=values}user/$fromExtn@Domain
> 'set:bLegVars=values,transfer:$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@gmail.com> wrote:
>>
>> how are you accomplishing that? by which technique?
>>
>> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
>> <fraserredmond@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@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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