search for: 777,1

...ar I've failed miserably - so I turn here for help. Basically I've been using the wiki and all the sample confs I could from there and via google. The queue system seems to work fine with my limited setup. Just 2 IAX2 clients where I keep Client B busy (by making it listen to mp3 via ext. 777) but logged into the queue. Client A then calls the queue (tried both ext. 7320 and 6320) and the announcements are fine ("you are next in line" etc.). When I make Client B not busy - it starts ringing like it should on the queue. But I never hear the MOH on Client A. Also - calling 777...

I'm having hard times with paging intercom Heres my dialplan exten => 777,1,Goto(intercom,777,1) [intercom] exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0) exten => 777,2,Page(Local/308 at page& Local/309 at page& Local/310 at page) [page] ; Paging context exten => _X.,1,Macro(page,SIP/${EXTEN}) [macro...

...hile blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid" extension: ======== exten => 777,1,Set(CALLERIDNUM=${CALLERID(num)}) exten => 777,n,GosubIf($[${LEN(${CALLERIDNUM})} != 10 ]?nocid,1:cid,1) ;prompt user for 10-digit #, and Return to GosubIf() exten => nocid,1,Read(CALLERIDNUM,/root/asterisk_sound_files/no_cid,10) exten => nocid,n,Verbose(User typed ${CALLERIDNUM}) ;Why...

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card application. when I call that extension, instead of getting " please enter you pin number" it fails and this is the output from th...

...p the channel. I tried either in php or in java but no success. In java i did this: ////////////// exec("Dial", "IAX2/400"); boolean t=true; while(t){ if(getChannel().getChannelStatus()==6)t=false; } wait(60000); hangup(); ////////// in my extension.conf: exten =>777,1,AGI(agi://localhost/sc.agi) the script is running but it does not hangup. Second solution,i tried this : exten => 777,1,Dial(IAX2/400,G(myscript)) exten =>777,n(myscript),AGI(agi://localhost/sc.agi) in sc.agi in this time i do not call exec("Dial","IAX2/400") statment;...

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called party, but the ringing sound continues and is heard over top of the conversation. If I add an Answer line to the dialplan, this problem disappear...

Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten => 777,1,NoOp(before) exten => 777,2,Dial(SIP/7|60|g) exten => 777,3,NoOp(after) priority 3 is never executed but this worked with Asterisk 1.0.7!!! TIA Giorgio Incantalupo

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten => _X.,n,Answer exten => _X.,n,NoOp(Ansage: Das Problem XYZ ist bereits bekannt und wird bearbeitet) ex...

Problem as follows: [default] exten => 777,1,Gosub(sub,1,1) exten => 777,n,Hangup() exten => h,1,NoOp(hung up in 'default' context) [sub] exten => 1,1,NoOp(in sub) exten => 1,n,Playback(tt-monkeys) exten => 1,n,Return() exten => h,1,NoOp(hung up in 'sub' context) This works fine if the caller listens to a...

...sion (mysip, 111, 2) exited non-zero on 'SIP/20-0889c4d8' -- Executing Dial("SIP/24-0889c4d8", "SIP/mediant/0") in new stack -- Called mediant/0 my extension.conf file is exten => 43,1,Answer exten => 43,2,Dial(SIP/43) exten => 43,3,Hangup exten => 777,1,Answer() exten => 777,2,Dial(SIP/777) exten => 777,3,Hangup() exten => 888,1,Answer() exten => 888,2,Dial(SIP/888) exten => 55,1,Dial(SIP/55) exten => 66,1,Dial(SIP/66) exten => _11.,1,Dial(SIP/mediant/${EXTEN:2}) exten => _11.,2,Congestion what is the problem -------...

...essage. I have read one of the post in the list which actualy show the URI string in the debug message (at the To: field). Is there any setting I need to set or turn on during compilation of asterisk? I have the head version of asterisk and my extension.conf setting is proveded below: exten => 777,1,Answer exten => 777,2,SetVar(VXML_URL=play=http%3A%2F%2Flocalhost%2Fvxml%2Findex.vxml) exten => 777,3,Dial(SIP/1234@192.168.1.72,10,t) exten => 777,4,Hangup SIP Debug message: *CLI> dial 777 -- Executing Answer("OSS/dsp", "") in new stack << Console...

Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ething nasty with the Record() function: If the user either hangs up during the prompt (ie. doesn't leave a message at all), or does leave a message but forgets to hit the # key at the end... Asterisk stops right there, so the rest of the script doesn't run: ======== [internal] exten => 777,1,Playback(leave_msg) exten => 777,n,Record(/tmp/test.wav,3,30) ;Verbose() not run exten => 777,n,Verbose(Here we are) ======== Am I doing it wrong? Is there a way to solve this? Thank you.

...so that it no longer blocks their number in outgoing calls. ====== Here's the command-line PHP script: #!/usr/bin/php <?php $fh = fopen('/root/output.txt', 'w'); fwrite($fh, "Received " . $argv[0]); fclose($fh); ?> ====== Here's extensions.conf: exten => 777,1,Set(CALLERIDNUM=1234567890) exten => 777,n,ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI(/root/dummy.php),${CALLERIDNUM}) ======... and here's the console :-/ -- Executing [777 at my-phones:1] Set("SIP/9001-088aa918", "CALLERIDNUM=1234567890") in new stack -- Exec...

...rno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out .= "UserName: admin\r\n"; $out .= "Secret: amp111\r\n\r\n"; fwrite($fp, $out); $in = "Action: ExtensionState\r\n"; $in .= "Exten: 777\r\n\r\n"; $in .= "Context: ext-did-custom\r\n\r\n"; $in .= "ActionID: 1\r\n"; //$in.= "Status: State\r\n"; $in .= "Action: Logoff\r\n\r\n\r\n"; fwrite($fp,$in); while (!feof($fp)) { echo fgets($fp, 256); //echo $fp; } echo "\r\n\r\n\r\n\r\n";...

...If we dial and outside number over this peer and then transfer the call into a MeetMe conference the Group gets decremented when it should not? This is most likely an error on my behalf, however I am not sure what the correct solution is. I have set the MeetMe conference up on a local extention 777. exten => 777,1,Answer exten => 777,n,MeetMe(9003|rpM) exten => 777,n,Playback(vm-goodbye) exten => 777,n,Hangup Do I need to do something in the above to preserve the Group count? Also there have been some complaints about callee's phone line being tied up and...

Hi I'm having trouble routing a call between two A*k servers I admin. SERVER- A: has a simple extensions set, and just needs to Dial to server B, but authenticate as part of the dial: exten => 777,1,Dial(SIP/abc-777:mypassword at someip.no-ip.info:5071/777,40,trw) exten => 777,2,Hangup So that should pass the call to the server listening on port 5071 of someip.no-ip.info, using the username of abc-777 and password of "mypassword", and pass it into extension 777 on that serv...

I set up the pilot number to voicemail to be 777. When a user calls 777 the voicemail answers and asks for mailbox, then password. Is there a way for the Voicemail to read what extension they are calling from and just ask for the password? I have a person complaining because they have to enter their mailbox number every time they check their v...

Hello all, Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies. So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10}) to determine which number is being dialed by the caller and then using a gotoif to get to