[Freeswitch-users] Installing OPUS codec

Wed Aug 20 20:57:34 MSD 2014

I installed from source as I mentioned mod_opus is in my source directory. But the mod_opus.so is not in my freeswitch installation module directory. So, for some reason the module didnt get build during my installation.  

From: freeswitch-users-request at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 98, Issue 134
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 19:54:30 +0400

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--Forwarded Message Attachment--
From: brian at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 10:47:48 -0500
Subject: Re: [Freeswitch-users] Installing OPUS codec

mod_opus is part of the default build, are you installing via packages or from source?

On Tue, Aug 19, 2014 at 7:57 PM, Kamrul Khan <dodul at live.com> wrote:

Hi, Can anyone tell me how to install OPUS module in FREESWITCH ? I have the source in my source directory. 		 	   		  

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--Forwarded Message Attachment--
From: brian at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 10:49:38 -0500
Subject: Re: [Freeswitch-users] Freeswitch srtp rejecting SRTP with 488

Show the logs on FreeSWITCH from right before the inbound invite arrives and after the call is rejected with 488, If you're doing WebRTC you have no reason to even be off on the SRTP wiki pages in the first place.

On Wed, Aug 20, 2014 at 5:43 AM, Daniel Ivanov <sertys at gmail.com> wrote:

He's implying that something else apart from the SRTP part is giving you the 488. Like an inability to negotiate a proper codec for the call.

On Wed, Aug 20, 2014 at 9:01 AM,  <dodul at live.com> wrote:

Hello,

Thanks for your reply. I'm sorry I guess didn't get you.

Are you saying the log I provided in my earlier email is incomplete?

Or I have to check some other log? If that is so can you please clarify which log? (Eg log name)

Or it is there in the log I provided, but I have ignored the part I needed to go through? If that's the case can you please indicate which part?

Thanks in advanced :)

Sent from my “contract free” BlackBerry® smartphone on the WIND network.

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Date: Wed, 20 Aug 2014 10:06:25

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Reply-To: freeswitch-users at lists.freeswitch.org

Subject: FreeSWITCH-users Digest, Vol 98, Issue 128

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--Forwarded Message Attachment--
From: andrew at cassidywebservices.co.uk
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 20 Aug 2014 16:53:58 +0100
Subject: Re: [Freeswitch-users] Freeswitch dialplan issue

secondly you may also need to set the actions as inline to use a variable you've just set. add inline="true" to you actions.

On 20 August 2014 15:03, Steven Ayre <steveayre at gmail.com> wrote:

Is the first condition matching?

On 20 August 2014 13:20, Nick Zaitsev <nick.zaitsev at mail.ru> wrote:

Good day to you.
I have strange issue.
I would like to set the ext in dialplan in that way:
if there is local registered ext in freeswitch fs should route the call there,else route the call outside, i wrote this in dialplan:

<extension name="fs3_outbound">
 <condition field="destination_number" expression="^([123][012345]\d\d\d)$">
 <action application="set" data="ext=$1"/>

 <action application="set"data="domain_name=${domain}"/>
 </condition>
 <condition field="${sofia_contact(*/${ext}@${domain_name})}" expression="error/user_not_registered">

 <action application="bridge" data="sofia/internal/${ext}@10.160.1.21"/>
 <anti-action application="bridge" data="user/${ext}@${domain_name}"/>

 </condition>
 </extension>
But it seems,that anti-action is not match at all(
Thank you for your time,best regards.
-- 
Nick Zaitsev

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