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...-04T16:29:27.254902Z 229 Execute SELECT * FROM queues WHERE name = 'cou0002-test' 2018-12-04T16:29:27.255606Z 229 Close stmt I also ran the query (SELECT * FROM queues WHERE name = 'cou0002-test') on the db and I do get a result. On Tue, Dec 4, 2018 at 9:08 AM Mitch Claborn <mitch_ml at claborn.net> wrote: > Maybe try capturing the queries that are executed on the mysql server? > That might point you in the right direction. > > -- show the log file name > SHOW VARIABLES LIKE 'general_log%'; > -- turn logging on and off > SET GLOBAL...

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

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.

Asterisk 16 latest DAHDI 3.0.0 latest Excerpt from chan_dahdi.conf is shown below. I'm trying to enable fax detection on inbound calls so that I can take appropriate action in the dial plan. "dahdi show channel 1" shows "Fax Handled: no". Does that mean that I don't have it configured correctly? [channels] ; Span 1: WCTE2/0/1 "WCTE23X (PCI) Card 0 Span

that was my first idea. and how should an other user know which number he should dial? user a: soft phone extension 100 hardware phone extension 101 On 06.02.19 15:25, Mitch Claborn wrote: > You can do this in the dial plan. Register the devices separately and > include both addresses in the Dial() command. > > > Mitch > > On 2/6/19 8:16 AM, basti wrote: >> In other words. >> >> I there a way that both phones are ring with only one ext...

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

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

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 "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

Short question: how to get control or notification (gosub, macro, AGI) when a queue member's phone starts ringing due to an incoming call from the queue. Backround: Our phone operators serve both an asterisk call queue and a queue for web chat support. I have a gosub on the queue that calls to our app server to mark the operator unavailable for web chat as soon as they answer an

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 11.1.0 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)

In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: > both phones are in the same net. > when the soft phone is shut down, on hardware phone only an led is > flashing to show an incoming call but no sound. > > both phones use the same extension. that is the reason why I use pjsip. > > On 06.02.19 14:59, Antony

Hi I am facing an issue where asterisk cannot see the queues that exist in my database through realtime. I am using res_odbc and a local mysql database. If I run: realtime load queues name myqueue I get "No rows found matching search criteria.", however if I do the same for a peer: realtime load sippeers name Then I get a result. Since my queues table is in the same database as my

...hould ensure that the bits you want set actually end up that way. As I understand it: Unix permissions = force_create_mode | (mapped_mode & create_mask) So use create mode to clear bits and force create mode to set them. Tim Allen ----- Original Message ----- From: "JW" <wilson@claborn.net> To: "Javid Abdul-AJAVID1" <AJAVID1@motorola.com> Cc: <samba@lists.samba.org> Sent: Thursday, December 06, 2001 6:07 PM Subject: RE: Adobe Photoshop uses wrong permissions when saving, default A CLs and create mask being ignored. > At 11:52 AM 12/6/2001 -0600, you...

At the end of the output for "core show channels verbose" is a line that reads "4 active calls". Does anyone know how that number is formatted if there are more than 999 active calls? Will it have a comma or not? -- Mitch

Is it possible (and advisable) to copy menuselect options from Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy? -- Mitch