<div dir="ltr"><div><div><br></div>Hi Stefano,<br><br></div>just to check the simplest of cases, use an extension like the following, and get a SIP trace (eg: from fs_cli: sofia global siptrace on)<br><br><div><div><br><extension name="calls_to_ivr"><br>
<condition field="destination_number" expression="^9999"><br>
<action application="set" data="bypass_media=true"/><br>
<action application="set" data="ignore_early_media=true"/><br>
<action application="bridge" data="sofia/internal/<a href="mailto:9000@10.1.1.1" target="_blank">9000@10.1.1.1</a>"/><br>
</condition><br>
</extension><br>
</div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 8, 2015 at 12:19 PM, stefano <span dir="ltr"><<a href="mailto:stefano.favaro@edistar.com" target="_blank">stefano.favaro@edistar.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
<br>
I'm trying to use "bypass_media_after_bridge" in my dialplan.<br>
In this scenario I have<br>
a) softphone client using g729<br>
b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit)<br>
c) Dialogic HMP 3.0 (g729 + g711)<br>
<br>
I'd like to use FS in passthrough mode so calls from a) will be rerouted to<br>
c).<br>
<br>
If I use the "bypass_media_after_bridge" signalling of calls is ok but i<br>
have no audio on a).<br>
<br>
If I use "bypass_media" audio is ok but I don't have the answer 200 OK back<br>
to my softphone a)<br>
<br>
This is my dialplan:<br>
<br>
<extension name="calls_to_ivr"><br>
<condition field="destination_number" expression="^9999"><br>
<action application="export" data="jitterbuffer_msec=60:200:40"/><br>
<action application="set" data="bypass_media_after_bridge=true"/><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="set" data="ignore_early_media=true"/><br>
<action application="set" data="effective_caller_id_number=1234"/><br>
<action application="bridge"<br>
data="{execute_on_answer=start_dtmf_generate}sofia/internal/<a href="mailto:9000@10.1.1.1">9000@10.1.1.1</a>"/><br>
<action application="pre_answer"/><br>
</condition><br>
</extension><br>
<br>
In my sip profile the "inbound-late-negotiation" param is set to true.<br>
<br>
Is there any mistake in my dialplan?<br>
<br>
Thanks,<br>
<br>
Stefano F.<br>
<br>
<br>
<br>
<br>
--<br>
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</blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature">Sincerely,<br><br>Giovanni Maruzzelli<br>Cell : +39-347-2665618<br></div>
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