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Greetings all- During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? Thanks! --Tim

...ILK codec and meet with some success on incorporating it in pjsip (an open source sip client). now i'm trying to do the same thing on Asterisk. any documentations, pointers, etc i should look into? any help is appreciated. -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20

In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>

I am using a context to change values in a DB. Currently in my context, I am passing it to exten => s,1,WaitExten(7) ; 7 seconds to input exten => s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only way I know how to 'grab' user input, which was normally from ${EXTEN} but I realize this won't work for extension 's'...... The short google search I did

How can I mute the message "please enter the number you wish to call and press the # key" in a2billing??? I tried use_dnid = YES but still I keep getting the message prompt... thanks

Still a couple of questions...... I did configure extconfig.conf ... ;iaxusers => odbc,asterisk ;iaxpeers => odbc,asterisk ;sipusers => odbc,asterisk sipusers => mysql,asterisk,sip_devices sippeers => mysql,asterisk,sip_devices ;sippeers => odbc,asterisk ;sipregs => odbc,asterisk ;voicemail => odbc,asterisk ;extensions => odbc,asterisk ;meetme => mysql,general

...x = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=defaultspan1 channel => 1-23 -- Edwin Lam <edwin.lam at officegeneral.com> Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

Hello List, I have found that asterisk supports only forwards in-dialog MESSAGE method. That is, if the MESSAGE method is sent within an active call. But according our requirement we need to send MESSAGE method to the other leg without being in a call (general stateless proxy forward). Is it possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran