similar to: app_queue dialback cdr problem

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings back to 111. Again at this point ext 111 isn't answered it

I made a post a few days ago with a patch to the queue to allow you to insert a penalty value in dynamic queue adding. http://asterisk.650dialup.com Since then, I decided I would make a whole new strategy called 'skillbased' I added a new option skillmask to the Queue appl and the AddQueueMember func as well as the queues.conf basicly if the skillmask of the queuemember is

Hi All, I'm stuck and need some help. I have installed the Asterisk CDR Analyser Version 2.0.1. It mostly works except for the CDR Report. I get the following error even though it lists the CDR details. Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day, sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE UNIX_TIMESTAMP(calldate) >=

Has anyone observed a problem where using Local channels with AddQueueMember results in missing TRANSFER events? Right now I'm using straight SIP channels when I call AddQueueMember(). I'm contemplating moving to Local channels because the non-state-based wrapuptime blows when you have a channel in multiple queues (they can hang up and get a call immediately so long as it's from a

Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122 at from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a

Both of our production asterisk servers are dumping core when making writes to our cdr tables. Here is a backtrace of the problems we are having: #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at util.c:347 347 if (tds_ctx && tds_ctx->err_handler) { (gdb) bt #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

Hi, I'm trying to use AddQueueMember() to add a member to a queue and trying to make this "logged" member in the queue between reloads and restarts of asterisk. I configure en queues.conf: [general] Persistentmembers=yes And Extensions.conf: exten=> *01,1,AddQueueMember(queue_name,Local/${CALLERID(num)}@default,penalty); When I log with

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257 #3 0x08092b3f in handle_reload (fd=33,

hi, our customer reports problem when 2 agents answer the call in the same time faster operator (device) answer the call, but the second is showed up (on device) and call is without sound asterisk 13.9/app_queue with strategy ringall/operators via Local channel with sip device (chan_sip) do you have any tips/info before i will dig deep into logs/debug? checked google&issues.asterisk.org

Hi. I'd like to get the call-id from a call (the *C-0000061c* part) to set in on a field on cdr's, like [Oct 8 16:09:37] VERBOSE[25701][*C-0000061c*] app_queue.c: SIP/12234-00006b18 is busy [Oct 8 16:09:37] VERBOSE[25701][*C-0000061c*] app_queue.c: Nobody picked up in 0 ms [Oct 8 16:09:38] VERBOSE[25843][*C-00000622*] netsock2.c: Using SIP RTP CoS mark 5 ... same =>

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

Before I report this as a bug (and get whacked with more bad karma), I'd like to make sure I'm understanding this "feature." I'm defining a queue with a couple of SIP phones as the memebers -- not agents. queue.conf allows you to set an exit context such that if set (and you use the "T" or "t" option) allows the caller or callee to transfer the call to

Hi, I've just uploaded a patch here: https://issues.asterisk.org/view.php?id=16925 This patch introduces a new parameter; "congestion" to both RINGNOANSWER in queue_log and AgentRingNoAnswer AMI event, which is set to 1 if the call failed to go through because of technical difficulties. And it also is more verbose than app_queue has been earlier, since app_queue usually silently

The Asterisk Development Team would like to announce the release of Asterisk 16.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello all. I'm setting up a new call center PBX using * v1.4, and figure it's better to go with AddQueueMember over AgentCallbackLogin. The functionality of AgentCallbackLogin still works, but without a firm idea of how long it will be in the codebase, I'm wary of building a system on top of it. The basic mechanics work, but I'm having some trouble on logging. With

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I know I'm missing something obvious, but I cannot wrap my wits around this one. I've been staring at it for too long I think. Maybe it's the three am syndrom! : ) So a call comes in and my snom ends up with this entry: CALLER NAME <sip:1231231234@server.ip> under missed calls, or whatever. Now I want to just click OK and

Hi, I am trying to get * log to mssql server. I have odbc and freetds configured, but my insert query is missing calldate which is a NOT NULL field in database schema. cdr_adaptive_odbc: Insert failed on 'sqlserver:cdr'. CDR failed: INSERT INTO cdr (clid,src,dst,dcontext,channel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid) VALUES