Robert Bielik skrev:
Ok, now pretty much everything is up 'n running, however when I try to send
an ANSWER (or any) command to *, it replies with
org.asteriskjava.manager.response.ManagerError Permission Denied. In
manager.conf for the *-java client, I have
read =
system,call,log
Ok, now pretty much everything is up 'n running, however when I try to send an
ANSWER (or any) command to *, it replies with
org.asteriskjava.manager.response.ManagerError Permission Denied. In
manager.conf for the *-java client, I have
read =
Moises Silva skrev:
You mean you cannot see AsyncAGI events? did you enable agi in the
read= parameter in manager.conf for your Java application user?
Yeay!! Thank you! No, I have not. And I suspected that I had to put something
there, I've googled mad for it
but have not found one document
Now that everything seems to rock I've hit the next hurdle. In my
extensions.conf I have the extension:
[agi-async]
exten = _01,1,Agi(agi:async)
and I can see that the context is hit when dialing into *. However my java
app that's supposed to receive
async agi events get no such events at
Someone? As * is used so extensively with SIP I must've made a _glaring_
mistake in my config (!)
/Rob
Robert Bielik skrev:
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what
Lacking any response I tried to set insecure=invite on both sides. And lo and
behold, the call
gets through.
Now, is this good or bad?
/R
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To
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
Machine 1
Ooops.. forgot. The versions of * are:
Machine 1: 1.6.1.4
Machine 2: 1.6.0.5
/Rob
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Hi all,
I've setup two * servers which are SIP interconnected ala osaka/toronto from
the * book (before anyone sugggests using
IAX instead, no, I NEED to have them SIP interconnected for verification/test
purposes). Then I have a
Zoiper connected to one of them via IAX (so that * will not
Hi all,
I'm trying to find some info on how to create my own dialplan
applications. Like f.i. Echo (ast_echo.c in apps). The API used in there
is what I would like docs on.
TIA
/Rob
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