Hi Jason. It seems your doing things right whatever that means. I
think the problem is more hardware related. Sure you have line in the
FXO?? have you tried dialing directly from some IP Phone?? I have
several applications that relay on automatic call generation with
Asterisk Manager and a PHP classes i have. But, as i said, i think the
problem is related to the configuration of the card. what does ztcfg
-vv says? what does zttool says??
best regardsOn 12/25/05, Jason D. Wolfe [EMAIL PROTECTED] wrote:
Hello,Somehow I've missed something here, so hopefully I'll be able to provideenough of my setup to get some help.I feel I'm very close to gettingit, but missing something none the less...1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked
to two POTS lines.2. I have the following entry in zapata.conf file:usecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechotraining=yescallprogress=no
context=incomingsignalling=fxs_kschannel=43. I have the following entry in extensions.conf[callAgent]exten=outbound,1,Dial(Zap/4/phonenumber) ;where phonenumber is a 10digit number
exten=outbound,n,Playback(access-code) ; just for the sake of doingsomething!4. I am using Asterisk Java Manager AGI OriginateAction with thefollowing code in a jsp page running on atomcat server:
//manageAGIManagerConnection managerConnection;ManagerConnectionFactory factory;OriginateAction originateAction;ManagerResponse originateResponse;factory = new ManagerConnectionFactory();
managerConnection = factory.getManagerConnection(192.168.1.4,jason,nosaj111);// connect to Asterisk and log inmanagerConnection.login
();originateAction = new OriginateAction();originateAction.setAsync(true);originateAction.setChannel(Zap/4);originateAction.setContext(callAgent);
originateAction.setExten(outbound);originateAction.setPriority(new Integer(1));originateAction.setTimeout(3000);originateResponse =managerConnection.sendAction
(originateAction, 3);6. when I execute the jsp page, I watch the console started with/usr/sbin/asterisk -cvvand I get the following message (I substituted phonenumber in for the 10digit number again)
*CLI == Parsing '/etc/asterisk/manager.conf': Found== Manager 'jason' logged on from 192.168.1.3 Channel Zap/4-1 was answered.-- Executing Dial(Zap/4-1, Zap/4/phonenumber) in new stack
Dec 25 10:55:40 NOTICE[3989]: app_dial.c:1010 dial_exec_full: Unable tocreate channel of type 'Zap' (cause 0 - Unknown)== Everyone is busy/congested at this time (1:0/0/1)-- Executing Playback(Zap/4-1, access-code) in new stack
-- Playing 'access-code' (language 'en')== Manager 'jason' logged off from 192.168.1.3== Auto fallthrough, channel 'Zap/4-1' status is 'CHANUNAVAIL'-- Hungup 'Zap/4-1'
exten = outbound,1,Hangup()What I eventually want to accomplish is the following:I want a web user (using a JSP page I think) to be able to click abutton and cause asterisk to dial outbound on both FXO ports, wait for
an answer, play some files, accept some input, and bridge the two callstogether.am I on the wrong track?is there anything that is standing out that Iam just not understanding here?ANY comments will be much appreciated.
Thank you,Jason___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
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