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-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Madushan Geethanga
Sent: 15 July 2016 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at 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.Alzola at
lebara.com<mailto:Joaquin.Alzola at 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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160715/ed36d7ef/attachment.html>
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
> 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.
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 .
> 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.