SearchSearch  Log in to check your private messagesLog in to check your private messages  recent posts Recent Posts
Post new topic   Reply to topic
View previous topic Printable version Log in to check your private messages View next topic
Author Message
herrcoreOffline



Joined: Oct 25, 2007
Posts: 2

Status: Offline
Posted: Oct 25, 2007 - 10:10 PM Reply with quote Back to top
I am developing a softphone based on the sip-communicator and java-asterisk API's. I was wondering if anyone could help me with sending a message to asterisk to mute/ unmute a channel?

I am using the Trixbox flavor of asterisk.

Thank you

Sergei
View user's profile Send private message
herrcoreOffline



Joined: Oct 25, 2007
Posts: 2

Status: Offline
Posted: Oct 29, 2007 - 07:49 PM Reply with quote Back to top
Incase anyone has a similar problem I found a simple solution is to simply stop the transmit RTP stream. You need to implement some extra methods as, the stopTransmit method in the mediaManager actually destroys the stream too. If you just do a simple process.stop() and process.start() inside the transmit class you won’t need to continuously re-create new streams. Simple and elegant, thanks all.

Sergei
View user's profile Send private message


View previous topic Printable version Log in to check your private messages View next topic

Post new topic   Reply to topic
Forum Rules and Guidelines | About VoIP User | Privacy Policy


All logos and trademarks in this site are property of their respective owner.
Comments and posts are property of the poster, all the rest (c) 2003-2008 VoIP User Limited.

VoIP User Limited is incorporated in England and Wales under Company Number 6694577.

No part of this site may be reproduced without our prior consent.