Hello, We are toying with setting up a redundant data center for our hosted PBX
product, and plan to use the OpenVPN feature of our Yealink phones for
connectivity to each data center. The feature has been fantastic with
the first data center, allowing us to bypass all SIP NAT issues entirely
and allowing remote access to the phones' web interface without having
to touch the customers' firewalls. We recently discovered that the Yealink platform only supports a single
tunnel, and have been discussing options with them. They asked if any
I'm trying to configure the dial plan for my company phones.
I'm seeing that there's a recommended dial plan of "X+^" and then check
the "send dial plan terminator" This seems to work ok. However I'm also seeing that a lot of people
completely specify their plan like this:
"1[2-9]xxxxxxxxx|[2-9]xxxxxxxxx|60[2-9]|6[1-2][0-9]|63[0-2]|70[0-9]|71[0-5]" Is there a benefit to specifying the full dial plan? Or should I just
use the smaller plan that matches everything? Thanks,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
(just like "dahdi show version", for instance). Though I'm currently asking this for spandsp, this question is on a
more general plan (for example, which ssl library am I currently using
?). Suggestions ? Regards
On Sat, Dec 3, 2011 at 12:59 AM, white hat
> When a caller calls my google voice phone number, I must answer, wait and
> press one to accept. Sometimes even that does not work.
> I just need a little advice on how to write the dial plan. I still have
> much to learn about asterisk, and appreciate any advice.
Geez, Maybe I am just brute forcing it, but, the following dialplan seems to
work (at least, most of the time!):…
When you are dialing a regular extension you might do something like this:
exten => Dial(123)
that would presumably dial extension 123.
but when one is using freepbx to admin the asterisk, and building a custom piece of dial plan code, how do you access a ring group?
exten => Dial (Local/601@xxxx)
how does one know what context to specify? and is the local keyboard needed? is Local case sensitive?
help is appreciated.
In the dial plan language of asterisk, what is the difference between prompting the user with a Playback() command vs. a Background() command? I want in a part of my dial plan to ask the user a prompt, and wait for 4 digits to be typed in. I don't want the user to have to end the string with a pound or something, just wait 2 seconds after they stop typing. ANd I do want the prompt to be interruptible if the user is fast and knows already what to do… I need to do some tests on the number…
If asterisk or freeswitch would be taught in a classroom environment,
is there someway to emulate and script emulation of users calling
in/receiving calls, transferring calls &etc? The plan is to have each student setup there own Asterisk or
FreeSwitch box, and measure handling efficiency, and communicate
between the servers (by transferring calls from server to server). Thanks for all suggestions, Alec Taylor
I wanted to get some input on what you all think is the best way to lookup
database data from asterisk dial plan. This is a two fold question.
1. I am using fun_odbc to pull settings and values back and it works good
but is there a better way. I want to maintain performance and simplicity as
much as possible.
2. func_odbc does not appear to allow for reading multiple records in a
return set. Is there a way around this or is there a better method.…