similar to: Queues and penalties

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

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using

Hi All I am trying to get voicemail switched over to ARA on version 13 and notice that the password is not stored in the db when it is changed. A little hair pulling and playing around and I think the problem is in the function ast_update2_realtime in main/config.c. Issued source is ==> int ast_update2_realtime(const char *family, ...) { RAII_VAR(struct ast_variable *,

Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names like 111.222 at mydomain.com. Using Kamailio as outbound proxy, it sends Asterisk a sip message where To header value is < sip:111.222 at mydomain.com> and From header has value "username" < sip:111.333 at mydomain.com;transport=UDP>;tag=fc609171. When Asterisk sends out the

Paddy, It's pretty easy to spot from the CLI. A voicemail gets called. And the screen basically stops scrolling from there. Eventually you'll get the "Task processors exceeded 500 queued tasks" or something like that. And maybe channels attempting to hangup due to lack of RTP (If you have no-rtp timers configured). Once you find the problem mailbox, You can call it via any

We ultimately found this to be a voicemail issue. The voicemail is held in MYSQL as well (via ODBC). And we found when attempting to playback a customers voicemail unavail greeting is when the deadlock would occur (Immediately, every time. Throwing the same "task processors" errors, And making pjsip completely unresponsive). We had imported a number of greetings from a legacy asterisk

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all &Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten =>

Hi all A quick question about busy lamps I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and go solid red when call gets answered but stay green when a call is made from the extension. Setup is Ext 200, 201, 202, each monitor the other two when 200 calls 202 - the BLF on 200 and 201 flash red - when 202 answers both 200 and 201 show BLF for 200 as red but

Hi List I have had a look at the various posts on this and seem to be more confused than ever - but then again that's not hard ;-) I am using Version 1.4.33.1 build from the Debian "lenny" distros I am trying to implement a simple screening [macro-screen] exten => s,1,Background(press1) exten => s,n,WaitExten(5) exten => 1,1,NoOp(accepted) ; Dont set a reply so dial

Hi List This may be a silly question by web searches etc don't seem to answer it. Is there a CLI command to display ALL channel variables - standard and user created - for a specific channel? something like show channel SIP/Test123 all I'm using Version 1.4.33.1 PG -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it

Howdy, have working realtime queues using queue_members looking something like; queuea|Local/101@context|0 queuea|Local/102@context|1 queuea|Local/103@context|10 Regardless of what strategy is used in the queues (roundrobin,rrmemory,ringall etc) it wont escalate on NOANSWER Asterisk SVN-branch-1.2-r33841 Any clues are appreciated! /Danny

Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0

Can anyone shed any light on an issue with agent penalties? I have 2 queues set up with agents working both queues, but where agent 1 should have a penalty for queue 2 and agent 2 should have a penalty for queue 1. When a call is sent to either queue, it rings agents with and without penalties at the same time. I set up a second system and cannot replicate the issue on the test system. I

Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done using penalty: queues.conf: member => SIP/dev1,0,Agent1 member => SIP/dev2,0,Agent2 member =>

Hi Paddy! This used to be 80 characters total (including all characters like channel type, '&' and '/'. Had the same issue in the past where I extended that in the code and recompiled. From what I understand there is basically no longer a hard limit in Dial since the recent change in the latest versions other than a single device must not exceed this but you can concatenate

On 6/5/2020 12:21 PM, Johnny Hughes wrote: > if you click on the six digit number, for example, e52775 for the > current latest "import 389-ds-base-1.3.10.1-9.el7_8". The result is > every diff of every change for the rpm. That's quite handy! But not what I'm looking for. I'm trying to figure out what edits I made to my config files. My most recent case was