Hi All,
I am trying to configure webRTC phone example for SIPml5 and i found this
info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
.
I have asterisk 11.9.0 installed and downloaded source of SIPml5 from
http://code.google.com/p/sipml5/source/checkout I copied sample co
On Fri, 09 May 2014 17:37:14 +0200
jg wrote:
> Either you do not compile the srtp module into the Asterisk package
> or you disable RTP encryption on a phone by phone basis.
Thank you for your help :)
> jg
>
dominique
--
Not that I'm aware of.
SIPAddHeader won't help you. Asterisk only sends the extra headers when you
use the Dial app.
You'll need to install a SIP Proxy in front of Asterisk if you want to
manipulate the SIP headers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[m
Hi,
Is there a way in Asterisk 11 to use a single voicemailbox for multiple
extensions while still hearing each extension's individual greeting?
Use case: someone has 2 numbers and wants all voicemail messages for
both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER
you w
Why don't you use the voicemail copy feature?
Create 3 mailboxes 1234, 6789 and 2000 for the shared.
VoiceMail(1234@default&2000@default,su)
VoiceMail(6789@default&2000@default,su)
Set both 1234 and 6789 to email the voicemail to a fake email address and
delete after email.
A copy of the message