similar to: Call Filtering

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

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: --

Greetings, I have a dialplan that calls the dictate application, but I want to do some post-processing on the RAW file created. The post processing is working fine as long as the dictation application exits gracefully, but fails when the user simply hangs up. How can I make sure the system() command is run regardless? Example: [test-dictation] exten => 123,1,Dictate(/tmp/dictate) exten

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

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm testing an ivr implementation (first time) using: exten => 620,1,Wait,1 exten => 620,2,Answer exten => 620,3,DigitTimeout,5 exten => 620,4,ResponseTimeout,10 exten => 620,5,Background(npi-greeting) ; "Thanks for calling press 1 for" exten => 1,1,Goto(npi-directory,s,1) For initial testing, I've arbitrarily mapped this onto ext 620 (will change that later

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

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include => client1_internal include => client1_outgoing_calls include => test_calls include => parkedcalls [client2_phones] include => client2_internal include =>

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

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hi everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150

Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: