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
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,verbose,agent,user,config,dtmf,rep
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 documen
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
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUB
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 mis
Ooops.. forgot. The versions of * are:
Machine 1: 1.6.1.4
Machine 2: 1.6.0.5
/Rob
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/m
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 --
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 reinv
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
___
--Bandwidth and Colocation provided by Easynews.com --
asteri
10 matches
Mail list logo
