Re: [asterisk-users] Async Agi problem

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

Re: [asterisk-users] Async Agi problem

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 =

Re: [asterisk-users] Async Agi problem

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

[asterisk-users] Async Agi problem

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

Re: [asterisk-users] SIP interconnection problem

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

Re: [asterisk-users] SIP interconnection problem

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

Re: [asterisk-users] SIP interconnection problem

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

Re: [asterisk-users] SIP interconnection problem

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:

[asterisk-users] SIP interconnection problem

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

[asterisk-users] dialplan applications

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 --