similar to: Notification when queue member's phone rings

Asterisk 11.1.0 I'm trying to use the "b" subroutine of the Dial application so that I can do some stuff with our internal applications that need to have access to the called channel information. I can see that the subroutine is being executed, but the arguments I pass don't see to make it to the subroutine. [callmenow] exten => s,1,NoOp(callmenow: Queue without answer)

hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number

Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will

I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

Asterisk 12.5 I have a reproducible segfault using the MeetMe application. How do I gather the necessary information (backtrace, core dump...) to submit a bug report? -- Mitch

Hi, I think this must be a bug introduced with 1.6.2.17.something. When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18, my AEL Dial() commands with the "U" options fail with the following error: [May 3 12:05:54] ERROR[6300] app_stack.c: Attempt to reach a non-existent destination for gosub: (Context:screen, Extension:s, Priority:1) Here are the segments

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a

Converting this customer from a MiTel system to asterisk. Discovered that the inbound calls from the T1 are going to extension 366. (This was mapped in the MiTel for some arcane purpose.) The dial plan I am currently using is shown below. When loading the dial plan, I get this warning: WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for an extension is strongly

Hi, i have a problem with Guild wars, it runs only in the backround. I ran it through terminal and got this: Code: fixme:win:EnumDisplayDevicesW ((null),0,0x32eb54,0x00000000), stub! fixme:win:EnumDisplayDevicesW ((null),0,0x32e6c0,0x00000000), stub! fixme:devenum:DEVENUM_ICreateDevEnum_CreateClassEnumerator Category {cc7bfb41-f175-11d1-a392-00e0291f3959} not found

The "core show channels verbose" command shows a "calls processed" value. Mine is currently 1928273. Exactly what does this figure represent? How is a "call" defined in this context? -- Mitch

Asterisk 12.4 I am seeing message "Error opening file for reading: Permission denied" several times during the asterisk startup (asterisk -cvvvvv) but it doesn't say which file. Is there a way to find out which file is having trouble? -- Mitch

Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit WARNING[5797]: presencestate.c:147 ast_presence_state_helper: No provider found for label CustomPresence ERROR[5797]: pbx.c:4375 ast_func_write: Function PRESENCE_STATE not registered I only see these when DPMA is enabled. Any ideas what causes this or how to correct it? -- Mitch

Asterisk 12.5 The CoreShowChannel event (in response to the CoreShowChannels action) no longer returns the "Application" field as it did in Asterisk 11. Is this a bug or a feature? -- Mitch

Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 "Wildcard AEX410" *53 FXO FXSKS

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

You might wish to read this thread as well, for some backround on LLVM IR. http://lists.cs.uiuc.edu/pipermail/llvmdev/2011-October/043719.html Summary: LLVM IR is target specific, not portable between different targets. LLVM IR is actually a Compiler IR and not a virtual machine language.

hi, i´m looking for an implementation of really low bitrates (16k per channel or less) with ogg vorbis. ome backround informations: i work on a portable device for streaming over the air with hard storage limits, so i like to reduce the filesize without loosing all quality. any ideas or suggestions? thanks oliver --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project