similar to: Executing commands even if user hangs up.

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

Hi, I am trying to set up a dial plan und I have a few problems to realise some functions. The dial plan should look like this: 123,1,Answer() 123,2,Queue(1stlevel,t) 123,3,Queue(2ndlevel,t) 123,4,Queue(3rdlevel,t) 123,5,Hangup() If a member of the 1stlevel-Queue can answer the call it should be hanged up after finishing. If not, the current member answering the call should be able to transfer

Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few

Hello One and All, I am a Linux admin, new to asterisk. I have been assigned the task of setting up a dictation server for the company I work for. Our company is into transcription. Currently we are using dictation server, which is provided by another company. Now we have decided to have our own dictation server. I have installed asterisk and have gone through many documentation and

Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create

I've been considering replacing a PRI with SIP or IAX trunks. The monthly cost difference is marginal, but it would save a bit on the hardware side and soft trunks would be easier to manage. I can't help but wonder what I would be giving up? I'd like to hear some "lessons learned" from those who are doing it or decided, for whatever reason, it's a bad idea.

Hi, I would like to develop a dialplan that allows the callee to reject the call like this:- 1) Call comes in and receives a greeting and get put into a queue. 2) A second call is placed to the member of staff (SIP phone or mobile phone) 3) The member of staff answers the call and is presented with a few options. 4) If the member of staff presses 1, the incoming call is connected to the member

[[apologies if this is a dupe. gmane is a bit odd sometimes]] I need some help on a handicap accessibility project. It's really great for people like me that naturally speaking is working in wine (mostly). One important shortcoming is getting our dictated text into linux. Today, in order to copy text from the wine environment and place it in the Linux environment, we need to dictate into

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Hi everyone I am having problems with my call queue We currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network operating center provide customer care services for customers who call in after the last

Hi there, i have a Problem with the Pickup command. Versions: asterisk 1.4.1 on gentoo my extensions.conf [only the interesting part]: [incoming_1] exten => 123,1,Ringing exten => 123,2,Dial(SIP/xxxx,20,r) exten => 123,3,wait(90) exten => 123,4,hangup [incoming_2] exten => 456,1,pickup(123@incoming_1) both are sip-accounts and have pickupgroup=1 in the sip.conf so my idea is,

Good day, does anyone have AstTAPI running for incoming calls, and would like to show some examples. My setting right now looks like this: sip.conf -------- [22] type=friend dtmfmode=info username=22 mailbox=22 secret=privat host=dynamic context=privat canreinvite=yes callgroup=1 incominglimit=2 extension.conf -------------- exten => 123,1,noop ;Hint(SIP/22) exten =>

Hello, My problem is the following : I have a Debian (testing distribution) on which I used to have Samba server v3.0.2a it worked perfectly with my other machine which is running Windows 2000. I recently made a global packages update on my Debian machine, which apparently changed the Samba server to 3.0.7 Since then, I have been absolutely unable to retrieve files from the debian server to the

Hello all, I have a very basic question but I haven't found any answer. I would like to configure asterisk so that it wil not indicate a call waiting to a SIP phone if it is already on conversation (off hook). But I don't want to loose call transfer, call hold and so on. Is there any possibility to do that? Regards, Daniel ANDRE -- Daniel ANDRE (mailto:daniel.andre@iris-tech.fr)

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Eric S. Johansson wrote: > vitamin wrote: > > > Eric S. Johansson wrote: > > > > > Is it possible to make this code (natlink) talk to naturally speaking in > > > wine using Windows 32 but speak to Python in Linux so we can do all our fun > > > command-and-control stuff there. > > > > > > > No, Linux's python has no notion of

A while back there was a thread in which many people were discussing how to make Netscape work with roaming profiles. There was some talk about how this would require a script to modify the registry, etc. I just talked to one of the NS programmers who said that the problem would be fixed and even enhanced in a version 4.5. He didn't say whether the 4.5 code was going to be an internal or

Eric S. Johansson wrote: > Is it possible to make this code (natlink) talk to naturally speaking in wine using Windows 32 but speak to Python in Linux so we can do all our fun command-and-control stuff there. No, Linux's python has no notion of COM (which is obviously a win32 only thing). And windows python doesn't work all that well on Wine.

Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf [default] exten