[asterisk-users] Asterisk and the media path

David Wessell david at ringfree.biz
Mon May 21 12:54:02 CDT 2012


So I need directmedia set in sip.conf on the LCR trunk.

1) Do I need it in the individual trunk settings for each pbx? Or is
in sip.conf enough?
2) Do I need anything on the pbx side that we are hoping to transfer media to?
3) How long into the call before the media is transferred over?

Thanks
David

On Mon, May 21, 2012 at 1:18 PM, Kevin P. Fleming <kpfleming at digium.com> wrote:
> On 05/21/2012 11:46 AM, David Wessell wrote:
>>
>> Hi Kevin,
>>
>> Thank you. Here's the requested information.
>>
>> 1) The Trunk is running 1.6.2.9. Also it's running a2billing.
>> 2) The PBX is running asterisk 1.8.12.0 along with FreePBX.
>> 3) I did directmedia on the trunk and canreinvite on the pbx since
>> they were different versions.
>
>
> Sure, but you used the *old* name for the option on the system running a
> *newer* version of Asterisk. That's why I was confused, I suspected you
> might have thought that 'directmedia' and 'canreinvite' were somehow
> different. Since both of your systems are 1.6.2.x or later, you can use
> 'directmedia' on all of them.
>
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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