It is not that easy to give the answer. There are lots of itsp typical
ways of registration and you haven't provide the info needed to help
you out.
You need a register line in the general part of sip.conf. It should
look something like (mine looks like this
register => ::@ipness.net:6060
If you add option 6 to the menu for the first position and use the
read command for the 2 last position and use a second line that looks
something like:
exten 6,n,Dial(SIP,6${ENTERED_NUMBER},20,t)
it should work.
The {ENTERED_NUMBER} should be the variable filled with the read
command.
D
Siax is working great for me and as far as I know/remember well, you
can get it from the app store for a reasonable price. It supports SIP
and IAX2 and works easy with Asterisk.
\erik
On 13 dec 2009, at 15:32, John Regal wrote:
> I have tried every sip phone offered on Apple's APP store (as
The easiest solution to deal with this is to have one context with
different extensions for the different numbers and route the incoming
calls from there. It should look something like this (not a tested
piece of asterisk script, just an example to give the idea).
Hope it helps :-)
Erik de
I used msmtp for delivering mail and this is the procedure I
documented once, based on info I found on the internet. I hope it is
of help.
Erik
Step 1 Installing needed packages/libs on your system
install this packages (I'm not sure if all the packages are needed but
with this packag
Siax is a pretty good working sip and iax2 softphone for the iPhone.
Easy to connect to your own Asterisk box
If you have an Android phone (I have HTC Hero with Android 1.5) ASip
is a good choice. It is working and and calls using umts are working
surprisingly well.
Erik
They are both av
I have read the posts about the security issue and from what I
understand there should be a check to make sure that the characters
used are actually allowed. I wrote a very straightforward and not so
rocket science kind of macro that will do the job I guess. Just two
parameters, one with th
achine and be requested to leave name
and phone number ;-)
Erik
On 17 feb 2010, at 00:41, Warren Selby wrote:
On Tue, Feb 16, 2010 at 4:38 PM, meetmecall
wrote:
Doesn't the built-in function FILTER() already do this?
*CLI> core show function FILTER
*CLI>
-= Info ab
GotoIfTime is about using the proper ranges of times, days of the week
and months and the months in the year. Check voip-info.org GotoIfTime
part. I think you need one line per month. The examples are clear.
Success.
Erik de Wild
Tripple-o
Your Asterisk migration partner
The Netherlands
Ve
Meetme() is the way to go. Running it on a virtual machine might not
be such a good idea bacause dahdi_dummy, needed for Meetme() might not
run. Google on "Meetme() cmd asterisk" and check the parameters
available. There is one for "listen only mode".
Don't forget to add a conference room to
Muhammad
It is not really your scenario but the scenario to setup a conference
call with three numbers could be to generate two call files that
points to a local channel/a context/extension that route the leg into
the conference room and have "your own leg" routed into the conference
room
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