similar to: AMI is not loaded

Hi guys, I hope you could help me. I am trying to get connected through AMI but something is not working. Both php code and manager.conf were working well in asterisk 1.4 1. Sometimes it gets connected and sometimes it doesn't: == Connect attempt from '192.168.25.241' unable to authenticate == Connect attempt from '192.168.25.241' unable to authenticate == Manager

Dear All, I have installed Asterisk 1.6.1.9 to use Bridge Application in AMI. After inatallation I have tried to connect the AMI via telnet. But it didn't connected. I used netstat to know the listening socket. But it was not available. How to start the AMI server socket. Please any one help me....... Thanks, Velusamy. -------------- next part -------------- An HTML attachment was

Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html

I am trying to connect Asterisk and Openfire together, but it's not yet working completely... I don't know for sure if my manager.conf-file is set correctly. I use this manager.conf file just to let Openfire talk to Asterisk... [general] displaysystemname = yes enabled = yes webenabled = yes (is this necessary for Openfire ???) port = 5038 bindaddr = 0.0.0.0 [openfire] secret=XXXXXX

hello friends, I am configurin my Billion ISDN and when I start asterisk (asterisk -vvvc) I have this error message: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 29 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line:

Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite ; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, "NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user

Hi, I have followed the all the steps given, in https://help.ubuntu.com/community/ActiveDirectoryWinbindHowto. to integrate the samba with active directory. I have the following configuration file, [global] workgroup = ASSURANCE security = ads realm = ASSURANCE.LOCAL encrypt passwords = yes winbind separator = + idmap backend = lwopen idmap uid =

Dear All, I have "get full variable" AGI call to get the ANSWEREDTIME channel variable. I have originated the call to one extension, once answered I have called DeadAGI to control the call. I have problem that after hangup the call AGI "GET FULL VARIABLE" returns -1 for ANSWEREDTIME channel variable. What is the problem? Where I made wrong. Please suggest me..

Dear all, In Sip.conf file how to setup incoming calls not to use authentication? Please provide some steps to do it.. Thanks... Regards, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090801/d6c39afc/attachment.htm

Hi. I have three servers running corosync and pacemaker, to maintain a floating address between them. This is working fine, and I can, for example, SSH to the floating address and get to whichever server has the address at the time. I am trying to connect to the same server (using the same address) for AMI, and it just isn't working, even though I can connect to the primary address of

Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. " WARNING[8446]: chan_oh323.c:3555

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Dear All, I have enabled the gatekeeper in oh323.conf file. I started the gatekeeper and also restarted the Asterisk. When I called, it was worked fine. After then enabled the routing mode in gatekeeper.ini file then I restarted the gatekeeper. When I called the routing mode didn't work. I have received the following warning in Asterisk console. " Jul 21 14:19:11

Hello, I just installed asterisk 1.6.1.1, asterisk-addons 1.6.1.1 and Asterisk-GUI 2.0 on Ubuntu 9.04 from *source. *Asterisk-GUI's web interface doesn't appear from this URL http://localhost:8088/asterisk/static/config/index.html*. *It says Not Found The requested URL was not found on this server. ------------------------------ Asterisk Server I already defined following config *in

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

Hi, I would like to get the status of asterisk extension with my php program. *My program as follows,* <html> <!--<meta http-equiv="refresh" content="1" />--> <?php $fp = fsockopen("xxx.xxx.xxx.xxx", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out