On August 2, 2012 during a window from 10:00AM to 11:00AM (Central Daylight Time, GMT-5), the core routers that provide connectivity through to all Asterisk community services will be swapped out. The actual service interruption may only last a few minutes, but could last longer. These services could be unavailable during most, if not all, of this time window. Once the move is complete, the services will be available again, with no user-visible changes. The services affected include: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org
I have a meetme running that is taking audio from a PC running asterisk
(console) as input
to my server that is then feeding it using meetme to two other asterisk
PC's going out the console.
All running 1.4.43 I have noticed that when the meetme first starts if I change the input
audio (new song) it changes very fast
at the output to keep up. Over time, like one hour later, if I change
songs it can be like 5 seconds before
the output changes. How can I keep that change…
I have two GV numbers. Both are configured to send calls to my Asterisk
18.104.22.168 box using the Google chat interface. At one time I had both
working with Asterisk. Now, for whatever reason, one of them has stopped
sending incoming calls to my asterisk box and instead just rolls to GV
voicemail. The other number continues to work fine. One is associated
with my wife's google account and the other is mine. I've compared our
account settings in Google and can't find any differences. Running "jabber
show connections" shows connections to each…
Please forgive me if I'm repeating this post.
I have searched and looked for similar question but have not seen a similar
one. When I do call from client to Asterisk 1.8, I do follows. 1. REGISTER
3. call established
5. call ended After the call ended, my client receive the message like this. "Really destroying SIP dialog 'email@example.com'
And at least after 30sec, my client disconnect from Asterisk.
I have to do REGISTER again. I think this message means that "Asterisk will…
I'm new to asterisk, currently working through Asterisk, The Definitive Guide. I have a couple Grandstream 200 phones and a Polycom 501 to play with. The Grandstreams were very easy to configure, but the more capable Polycom almost drove me crazy. There is so much conflicting information on the web about necessary config files and names of those files. Anyway, all my phones are now working on a basic level, but it occurs to me that once spend a lot of time learning Asterisk, I might then have to spend an equal amount of time figuring out how to configure…