similar to: queue need very long can start music

Hi everyone, I have encountered a hard problem that when i connect my anology phone to channelbank ,I found that i dial a number and create the call,then ,I hangup the call,but ,very quickly,I listen the ringing im my phone,I pick it up ,and found it noting, anybody can tell me this reasons,and how to solve it,Thanks! -- Best regards! jordan pan Location:Shenzhen China

Hi all , In my pbx ,there is no zaptel card ,so i loading ztdummy,but problem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

Dear R fans ( and wiki fans), I am just writing a draft to introduce confidence intervals of various "effect sizes" to my students. Surely, I'll recommend the package MBESS in R. Currently, it means I have to recommend R's interface at first. As a statistics teacher in a dept of psychology, I often have to reply why not to teach SPSS. Psychologists and their students hate to

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Dear LLVM user and developer, We get a chance to give a tutorial on LLVM and MCLinker. The tutorial will be co-located with CGO 2013 on Feb. 24 (Sunday morning) in Shenzhen, China. If you are also interesting in these topics, welcome to join the tutorial! Here is a website of the tutorial: http://code.google.com/p/mclinker/ We're also looking for additional presenters to share a

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 Asterisk users, I'm one of Asterisk ARI users, and trying to designing the new ARI for application execution in Stasis(). This will be made possible for executing the applications in the Stasis() application. But, before going further, I would like to know which application needs to be considered. Because this feature will introduce new Stasis behavior, I would like to test the

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

## Question1: How to define IV with interaction alone, without main effects? ## Question2: Should Type III ANOVA in package car be independent of the factor level order? ## data from http://www.otago.ac.nz/sas/stat/chap30/sect52.htm drug <- c(t(t(rep(1,3)))%*%t(1:4)); disease <- c(t(t(1:3)) %*% t(rep(1,4))); y <- t(matrix(c( 42 ,44 ,36 ,13 ,19 ,22 ,33 ,NA ,26 ,NA ,33 ,21 ,31 ,-3 ,NA

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

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

Hi all, does anybody know any way that when it run "reload app_queue" in the asterisk cli it don't lose the informations from "show queue (queue name)" ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command "show queue (queue name)" show me

About two weeks ago there was a thread about asterisk suddenly dying - I posted a response that the same happens to my asterisk about once a month, sometimes more. Someone suggested using 'safe_asterisk' (and get hold of a core dump) which sounds like a good idea, but one thing I can't figure is how to get "module reload app_queue" executed automatically at startup?

Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

--- Jonathan Tew <jonathan@ultracart.com> wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing

Guys, this error has been driving me nuts and I see no indication anywhere as to what it may mean. Anybody has any clues on this? -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') -- Playing 'vm-review' (language 'en') -- Saving message as is -- Playing 'vm-msgsaved' (language 'en') Mar 4 21:02:06