no audio with bypass_media_after_bridge
Hello, I'm trying to use "bypass_media_after_bridge" in my dialplan. In this scenario I have a) softphone client using g729 b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit) c) Dialogic HMP 3.0 (g729 + g711) I'd like to use FS in passthrough mode so calls from a) will be rerouted to c). If I use the "bypass_media_after_bridge" signalling of calls is ok but i have no audio on a). If I use "bypass_media" audio is ok but I don't have the answer 200 OK back to my softphone a) This is my dialplan: <extension name="calls_to_ivr"> <condition field="destination_number" expression="^9999"> <action application="export" data="jitterbuffer_msec=60:200:40"/> <action application="set" data="bypass_media_after_bridge=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="effective_caller_id_number=1234"/> <action application="bridge" data="{execute_on_answer=start_dtmf_generate}sofia/internal/9000@10.1.1.1"/> <action application="pre_answer"/> </condition> </extension> In my sip profile the "inbound-late-negotiation" param is set to true. Is there any mistake in my dialplan? Thanks, Stefano F. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-audio-with-bypass-media-aft... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi Stefano, 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) <extension name="calls_to_ivr"> <condition field="destination_number" expression="^9999"> <action application="set" data="bypass_media=true"/> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="sofia/internal/9000@10.1.1.1"/> </condition> </extension> On Wed, Jul 8, 2015 at 12:19 PM, stefano <stefano.favaro@edistar.com> wrote:
Hello,
I'm trying to use "bypass_media_after_bridge" in my dialplan. In this scenario I have a) softphone client using g729 b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit) c) Dialogic HMP 3.0 (g729 + g711)
I'd like to use FS in passthrough mode so calls from a) will be rerouted to c).
If I use the "bypass_media_after_bridge" signalling of calls is ok but i have no audio on a).
If I use "bypass_media" audio is ok but I don't have the answer 200 OK back to my softphone a)
This is my dialplan:
<extension name="calls_to_ivr"> <condition field="destination_number" expression="^9999"> <action application="export" data="jitterbuffer_msec=60:200:40"/> <action application="set" data="bypass_media_after_bridge=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="effective_caller_id_number=1234"/> <action application="bridge" data="{execute_on_answer=start_dtmf_generate}sofia/internal/9000@10.1.1.1 "/> <action application="pre_answer"/> </condition> </extension>
In my sip profile the "inbound-late-negotiation" param is set to true.
Is there any mistake in my dialplan?
Thanks,
Stefano F.
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-audio-with-bypass-media-aft... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
participants (2)
-
Giovanni Maruzzelli
-
stefano