similar to: diagnostic info for a segfault

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 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 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.

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

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 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

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 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)

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

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

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

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

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

Asterisk 11.1.0 running on Ubuntu 12.04 64 bit Dahdi Digium T1 card Occasionally, I will find an inbound call that just seems to be stuck, usually in our after-hours menu portion of the dial plan. This morning I had this one core show channels concise

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

Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. -- Mitch