Re: [asterisk-users] VoiceMail Audio playing

2016-07-15 Thread Joaquin Alzola
> No.  The VoiceMail server takes care of all that itself; it delivers the 
> broadcast and records the messages.

Thanks AJ.
This email is confidential and may be subject to privilege. If you are not the 
intended recipient, please do not copy or disclose its content but contact the 
sender immediately upon receipt.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


Re: [asterisk-users] VoiceMail Audio playing

2016-07-15 Thread A J Stiles
On Friday 15 Jul 2016, Joaquin Alzola wrote:
> Hi Madushan
> 
> Maybe I was not clear …. After SIP negotiation and SDP set up on the
> VoiceMail Server ….
> 
> Is there  a file to specify a MGw (the machine that deliver RTP packages to
> end user)?

No.  The VoiceMail server takes care of all that itself; it delivers the 
broadcast and records the messages.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VoiceMail Audio playing

2016-07-15 Thread Joaquin Alzola

> Asterisk does not separate things like this. For media originating from it 
> the source will always be it. That is if you do a SIP call to Asterisk then 
> media will come from that same Asterisk.

Joshua ok perfect so Asterisk already have the play module incorporated.
That’s great to hear so no need to integrate it to a MediaGatwey or SBC.
This email is confidential and may be subject to privilege. If you are not the 
intended recipient, please do not copy or disclose its content but contact the 
sender immediately upon receipt.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VoiceMail Audio playing

2016-07-15 Thread Joshua Colp

Joaquin Alzola wrote:

Hi Madushan

Maybe I was not clear …. After SIP negotiation and SDP set up on the
VoiceMail Server ….

Is there a file to specify a MGw (the machine that deliver RTP packages
to end user)?


Asterisk does not separate things like this. For media originating from 
it the source will always be it. That is if you do a SIP call to 
Asterisk then media will come from that same Asterisk.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VoiceMail Audio playing

2016-07-15 Thread Joaquin Alzola
Hi Madushan

Maybe I was not clear …. After SIP negotiation and SDP set up on the VoiceMail 
Server ….

Is there  a file to specify a MGw (the machine that deliver RTP packages to end 
user)?

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Madushan Geethanga
Sent: 15 July 2016 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] VoiceMail Audio playing

Hi,
VoiceMailMain is used to retrieve voice mails

http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain
Best Regards,
Madushan

On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzola 
<joaquin.alz...@lebara.com<mailto:joaquin.alz...@lebara.com>> wrote:
Hi Guys

Which module on Asterisk is the one in charge of playing the VoiceMail Server 
Audio to the end customer?
I have work with MRFP but is it a module included in the SW? Need and external 
source?

BR

Joaquin
This email is confidential and may be subject to privilege. If you are not the 
intended recipient, please do not copy or disclose its content but contact the 
sender immediately upon receipt.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

This email is confidential and may be subject to privilege. If you are not the 
intended recipient, please do not copy or disclose its content but contact the 
sender immediately upon receipt.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VoiceMail Audio playing

2016-07-15 Thread Madushan Geethanga
Hi,

VoiceMailMain is used to retrieve voice mails

http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMailMain

Best Regards,
Madushan

On Fri, Jul 15, 2016 at 3:07 PM, Joaquin Alzola 
wrote:

> Hi Guys
>
>
>
> Which module on Asterisk is the one in charge of playing the VoiceMail
> Server Audio to the end customer?
>
> I have work with MRFP but is it a module included in the SW? Need and
> external source?
>
>
>
> BR
>
>
>
> Joaquin
> This email is confidential and may be subject to privilege. If you are not
> the intended recipient, please do not copy or disclose its content but
> contact the sender immediately upon receipt.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users