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....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 general_log=&...

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

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 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

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)

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 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

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

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

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

Trying to set some CDR fields in the "connected" macro of a queue command. None of the custom fields I set are stored in the database, but I can set "userfield" and it does get set. I think that the macro runs on the agent's channel, not the caller's, and this might contribute to the problem. From the sample below "userfield" (and its alias operatorid)

asterisk 11.1 Documentation in cdr.conf for endbeforehexten reads: Normally, CDR's are not closed out until after all extensions are finished executing. By enabling this option, the CDR will be ended before executing the "h" extension and hangup handlers so that CDR values such as "end" and "billsec" may be retrieved inside of of this extension. I have

I read somewhere that DPMA is not supported for Asterisk 12. Can anyone confirm or deny that? If not supported yet, will it be? If so, when? -- 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

Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit [2014-08-21 16:37:49] WARNING[5797]: phone_users.c:5236 set_and_process: User SIP settings missing or invalid I'm getting the error message above when DPMA is enabled, using a fresh build but with my config files from Asterisk 11. Any idea what it means? I can't find the "phone_users.c" file to examine the source