directmedia

[danardf]

Hi all.

It's not often that i'm asking some help. But i've some doubts.

I'm using Elastix as IVR connected to an IPBX.
Some branch provides a menu to be redirected to a service. (So here, a phone number into this IPBX).
The problem is, i have not too many SIP channels, only 10, and i would like to use directmedia for redirecting the RTP and left no SIP channels after the redirecting calls.
I had a soluction but this one doesn't work.
I my custom diallpan, i launched a : tranfer(my_extension). But i had some problem when the CID number was empty. from the client. (I don't know why). Transfer failure,.because i had a busy return.

Here, i used goto(outbound-allroutes,my_extension,1) and it works fine, But i would like to be sure to redirect the calls and left all SIP channels after.
How could i use goto(outbound-allroutes,my_extension,1) and directmedia?

 

[Amphibian]

danardf,

You ask "How could i use goto(outbound-allroutes,my_extension,1) and directmedia?"

I believe that all you need to do is enable directmedia (directmedia=yes) and with your goto code all your calls should pass through to the IPBX and drop out your Elastix Server thus freeing up your SIP channels on your Elastix Server.

For asterisk 11 the configuration options are:
* directmedia=yes/no to enable/disable direct media per SIP peer

* a directmedia=nonat option to only allow direct media if a media NAT
is not in effect.

* a directmedia=outgoing option to only send direct media reinvites on
outgoing call legs (used to mitigate reinvite glare scenarios)

* a directmedia=update option to send SIP UPDATE requests instead of
reinvites in order to set up direct media.

* direct media ACLs to prevent a peer from using direct media if they
have registered from an address where direct media would not be
permissible/possible.

I haven't played much with it so others may have a better answer. I know there are/where issues with directmedia and NAT and that was why I didn't do much with it.

amphibian

 

[danardf]

Hi Amphibian.

How are you ?

Thanks for your help.
I will check again my configuration and will try again. :wink:

 

[Amphibian]

danardf,

I be good. We must do call again soon.


Have a great day,

amphibian

 

[sman]

Keep in mind that with directmedia neither end can be behind NAT as far as I know. If your PBX or your extension is behind NAT it won't work. I think you may be able to get around that on the extension side if you configure a STUN server but I've never been able to get that to work either.

Also, if you use SIP trunks the provider must support directmedia and a lot of them do not.

Also again, if any transcoding is going on it won't work.

 

[danardf]

Hi

Thanks for this advise.
I know this way is restricted
No problem for that, i doesn't use any NAT.