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....org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to be a real shortcoming in app_queue. Any ideas, suggestions, anyone want to work with me to sort this ? Paddy _____ From: John Kiniston [mailto:johnkiniston at gmail.com] Sent: 28 November 2018 21:17 To: paddy at wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues and penalties This should work, How are you defining your timeouts in the queues.conf ? And to verify, in your extensions.conf you are calling Queue with the queue name and the ruleset to apply fr...

...search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum string length for the dial data in the DIAL command Paddy _____ From: Dovid Bender [mailto:dovid at telecurve.com] Sent: 01 May 2020 10:26 To: paddy at wizaner.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Length of dial string 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] Ex...

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 All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

...;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 => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117 On Fri, May 1, 2020 at 3:22 AM Paddy Grice <paddy at wizaner.com> wrote: > 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> ] - >...

...is no unavail greeting. Each case deleting the problem recording from the database fixes the issue, And subsequent recordings for the same mailbox have no issue. *Nick Olsen* Network Engineer Office: 321-408-5000 x103 Mobile: 321-794-0763 On Wed, Apr 1, 2020 at 9:04 PM Paddy Grice <paddy at wizaner.com> wrote: > Hi All > > This sounds just like a problem I have had and still investigating having > moved to 16.9 using chan_sip. I am still trying to repeat the problem it > looks from debug that the issue is either voicemail of call transfer but I > cant consistently repeat...

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 *,

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

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

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

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olli Heiskanen Sent: 03 January 2015 08:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk removes a charachter from sip peer name Hello all, Just wondering on a behavior I noticed while testing with realtime sip peers with names

Hi All I have a problem with queues that I have been trying to solve for many months - the customer has now picked back up onto this and wanting a solution - any guidance, ideas or solutions welcome. This is the situation :- We have a number of agents in a ringall group, a call joins the queue and all handsets ring - great. An agent answers and handles the call - great. The call

Hi there I hope someone can help - I am having a big problem getting calls cleared from several asterisk systems when RTP timeouts occur. It appears that asterisk doesn't send a BYE when it decides to terminate a call because of a RTP Timeout - is this a configuration problem? if so what need changing? or is this a bug/feature? if so is there a work around? The setup here is calls come from

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 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

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 All This sounds just like a problem I have had and still investigating having moved to 16.9 using chan_sip. I am still trying to repeat the problem it looks from debug that the issue is either voicemail of call transfer but I cant consistently repeat it. Voicemail is using ODBC and I just imported the data from the old system into the new database. Nick - if you have any more info I

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