similar to: AMI and ExtensionState command returning bogus 'status' number

Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new

New installation from AsteriskNow 3.0.0 with asterisk 11 and freepbx, running Digium D40 and D70 phones. Direct-dialed extensions work fine, but extensions in RingGroup won't ring - dialparties.agi apparently removes them from the dialstring pre-emptively: dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN) dialparties.agi: Extension 2010 has ExtensionState: 4 What can I do? Any extension

I'm not having any luck getting the ExtensionState action of the Manager API to work. The response is always success but the Status is always -1 which to me means an error. Here is a typical telnet session. ----------------------------------- stockholm:~ # telnet localhost 5038 Trying ::1... telnet: connect to address ::1: Connection refused Trying 127.0.0.1... Connected to localhost. Escape

This is my first attempt to write software of any sort. What I am trying to is to use a .php page to query asterisk Manager and get the ExtensionState for each particular extension. Then when it has the answer it outputs an XML file for use as the directory on a Cisco 7960 phone. What I am thinking is that when the user hits the directory button to veiw the directory that is at this URL it will

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

In 1.4.43 I would see things from "core show channels" like DAHDI/18/xxxxx for line 18 in Asterisk 11 its DAHDI/i4/xxxx How do I get the line number back? Jerry

Can you then inform me on what structures this information is stored in, in the asterisk code? Then ill try to do a quick dirty version of the replication. Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Kevin P. Fleming Sendt: 9. juni 2006 16:25 Til: Asterisk Users Mailing List - Non-Commercial

I've got hints setup for my MeetMe conferences like so: exten => _60X,hint,MeetMe:${EXTEN} and they show up in "core show hints" like so 600 at dialtone : MeetMe:600 State:Unavailable Watchers 1 _60X at dialtone : MeetMe:${EXTEN} State:Unavailable Watchers 0 I'm wondering why they're Unavailable instead of

Hello all Has anyone had any success with the Manager API ? I am trying to check an extension status without too much luck I have the following <?php $fp = fsockopen("127.0.0.1", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out .=

Hello, For each incoming or outgoing call, sip hardphones I'm using, turn BLF on and off like this: the first call (after leaving idle status) turns 1st BLF on, the second one turns 2nd BLF and so on, when a call is hanged, its BLF is turn off. My first question is : do you think such behaviour is general ? My 2nd question is : using AMI, how can I tell for a given extension : 1. the number

Hello, For each incoming or outgoing call, sip hardphones I'm using, turn BLF on and off like this: the first call (after leaving idle status) turns 1st BLF on, the second one turns 2nd BLF and so on, when a call is hanged, its BLF is turn off. My first question is : do you think such behaviour is general ? My 2nd question is : using AMI, how can I tell for a given extension : 1. the number

Is there a limit to how many manager commands you can send pipelined in a single Asterisk management session? Right now it seems that if we dump a bunch of requests like this (two isn't enough though): action: ExtensionState exten: 906 actionid: foo-141489356-52 context: Hints action: ExtensionState exten: 897 actionid: foo-141489356-53 context: Hints the manager connection is reset by

Hi, How to know the all SIP extensions status with AMI's ExtensionState ? What is the value should I pass in Context: <> ?? which will be define at context here ? shell I use sip.conf's context for that extension or any other? extension : <> ?? extension will be SIP/100 or just 100 ?? Please guide me ........... ----- Thanks and regards Virendra Bhati +91-9172341457

Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker

Is it just my box, or is there something flaky in the implementation of "show manager commands"? Note: I'm using putty. About half way through this, I toggled my KVM over to the desktop and logged in to try and recreate it. The output was the same as the last two entries in this dump. bebop*CLI> show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager

After installing a new box and asterisk. i have got these errors [root at localhost ~]# asterisk Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory [root at localhost ~]# asterisk -vr Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) I didn't find a folder called asterisk in the directory /var/run [root at

Hi, I'd like to implement the following: Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to a custom IVR that says something like "extension $EXTEN is $DIALSTATUS. Please try again later or dial $CODE now to notify you as soon as $EXTN is available.". So the "notification" part is what I'm trying to figure out. The extensions are SIP (but

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi list! I recently upgraded to FreePBX 2.1.3 although I am not sure if this has something to do with it. I do a nightly restart of Asterisk, just in case. This has been working fine months but since a few days asterisk seems to die and I am not able to restart it again, I keep getting a socket in use message. This is on Asterisk 1.2.12.1, Zaptel 1.2.9.1 and Libpri 1.2.3 This is a snippet