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.)<br><br>Yes, the bridge doesn't start until after the A-leg has answered.<br>
<br>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.<br>
<br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
are you setting ignore_early_media=true in the first vars=values area?<br>
<br>
This looks like you could be calling one leg who is still not answered<br>
and then bridging it to another dest. The bridge app will wait for<br>
the first leg to answer before bridging.<br>
<br>
Also if you have any NAT anywhere, look for an "auto-changing port"<br>
type message which can also be attributed to this due to a detection<br>
period for incorrect ports.<br>
<br>
<br>
<br>
On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond<br>
<div><div></div><div class="h5"><<a href="mailto:fraserredmond@gmail.com">fraserredmond@gmail.com</a>> wrote:<br>
> event_socket:<br>
> api originate {vars=values}user/$fromExtn@Domain<br>
> 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline<br>
><br>
> then<br>
><br>
> dialplan:<br>
> <action application="set" data="effective_caller_id_number=+1800number"/><br>
> <action application="set" data="effective_caller_id_name="/><br>
> (set and/or export a bunch of other vars too)<br>
> <action application="set"<br>
> data="dial_string=sofia/gateway/<a href="http://gatewayname.com/00$%7Bdestination_number%7D" target="_blank">gatewayname.com/00${destination_number}</a>"/><br>
> <action application="bridge" data="${dial_string}"/><br>
><br>
> Cheers,<br>
> Fraser<br>
><br>
><br>
><br>
><br>
> On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale<br>
> <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>><br>
>> how are you accomplishing that? by which technique?<br>
>><br>
>> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond<br>
>> <<a href="mailto:fraserredmond@gmail.com">fraserredmond@gmail.com</a>> wrote:<br>
>> > The call is originated from Freeswitch (via CLI) to a softphone, then<br>
>> > when<br>
>> > that is connected it bridges out to the gateway.<br>
>> ><br>
>> > Cheers,<br>
>> > Fraser<br>
>> ><br>
>> ><br>
>> ><br>
>> ><br>
>> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale<br>
>> > <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
>> >><br>
>> >> Where is the other side of this call coming from?<br>
>> >><br>
>> >> [ ( ) ] -> FS -> (PSTN via SIP)<br>
>> >><br>
>> >> What goes in the empty space above?<br>
><br>
><br>
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