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
On Fri, 09 May 2014 17:37:14 +0200
jg webaccounts...@jgoettgens.de 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
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
Why don't you use the voicemail copy feature?
Create 3 mailboxes 1234, 6789 and 2000 for the shared.
VoiceMail(1234@default2000@default,su)
VoiceMail(6789@default2000@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
