similar to: Lock extension from asterisk

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

DISA is currently disconnecting when I dial 8888 to access DISA. Below is my extensions.conf file from A@H and some lines which shows the disconnect. Should DISA be loaded as a module in modules.conf? When I do a 'show applications' i see that DISA is there. Help! -------------------------------------- ;Asterisk CLI as I placed a call from cell into the system. Playing

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

All the aforementioned techniques need change everytime on the dialplan. I need the office secretary to edit a file (call file) and place it in a particular folder in their windows PCs. this folder is the outgoing folder of LINUX shared through samba in LAN. i need to make it as easy as possible, please. On Tue, Mar 20, 2018 at 5:41 PM, Frank Vanoni <mailinglist at linuxista.com> wrote:

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

I've been trying to use the VMAuthenticate function in 1.2+. This function is supposed to "behave[s] the same way as the Authenticate application, but the passwords are taken from voicemail.conf." The problem is that it always gives the "comedian mail" prompt and requests the mailbox number, even though I provide the mailbox number already. The upshot is that

I have a rather technical question here. I'm looking at the code in app/app_voicemail.c, I'm wondering when the vmauthenticate() function is called. Aside from being called by load_module() as follows: res |= ast_register_application(app4, vmauthenticate, synopsis_vmauthenticate, descrip_vmauthenticate); I can't see any other calls to it. Can someone explain to me at what point in

I want to authenticate a user before he is able to use the phone. I also want to set his privilege as to where he is allowed to call to... Preferably, the password should be their VoiceMail password, (every extension (or is that user?) can have voicemail defined - even if its not in use?) ...one should be able to enter the password (variable length) as part of the dial sequence - eg the number

Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use

Hi List, We have a remote asterisk SIP phone at the cottage. I'd like it to have minimal privileges when it first registers with Asterisk. Ideally it should be in a restricted context. Dialing any number would intercept the call and tell the person to log on. This way, if the phone was stolen or someone got into the cottage, we wouldn't have a bunch of surprise charges on our phone

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

Hellos, My voip carrier has assigned me a IP based account...where they only give me the IP to call through. I have setup the dial plan exten => _7XXX.,1,Answer() exten => _7XXX.,2,vmauthenticate(${CALLERID(number)}) exten => _7XXX.,3,Dial(SIP/${EXTEN:1}@Y.Y.Y.Y) exten => _7XXX.,4,Hungup() Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error SIP/Y.Y.Y.Y-35dc

GUys.. I dont know if this is a known bug or not but I just tested and replicated this one over and over again. It involves call transfer from calls that entered the pbx via a queue.. say a call comes in and its thrown in a queue, somebody answers the call but then wants to transfer the call to somebody else outside the queue, of course... the bug comes in here.. Im using mixmonitor to record

Hi guys This is my Ist post on this group. Is there any variable like ($VM_CALLERID for voicemail mailbox) for accessing Asterisk Voicemail password which is set through comedian mail.?????????????? plz reply me as soon as possible.... <html><div><PRE class=quote><IMG height=2 src="http://graphics.hotmail.com/greypixel.gif" width="100%"

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi Carlos " Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from 1.6 so you now need to use Dynamic Agents. Although they claim that is is simple enough to replace that functionality with dial plan code I have yet to see a one line example that replaces everything the agentcallbacklogin command did.| I totally agree, I have never seen any example that makes it work.

I did a cvs update this afternoon and since then asterisk doesn't seem to clean up the channels after they hangup. This has been working perfectly for quite some time previously... I do a show channels and it shows the channels still up. The only way out is to kill and restart asterisk.... I am frantically trying to would out how to get a non-current CVS copy of the source and get it back

When I try and start asterisk I get the following, however I have commented out the data the connections in res_mysql.conf and res_pgsql.conf. I am not sure therefore why I am getting these errors. Do I have to change something else to turn this off? Thanks Asterisk 1.4.21.2~dfsg-3, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk