similar to: This is a test, hijack this

Hi all, I'm testing the persistent dynamic queue members functionality on 1.6.0.10. The queue members are agents defined in the agents.conf file. When I issue an asterisk restart and check the queue members again on the CLI, all of them are listed as /invalid/ and there is no way to change this but to unload app_queue.so and load it again. My guess is that the internal AstDB queue

Hi, I didn't notice on my first answer, but we are on the -dev list and this is not related to asterisk code developing. I will answer you on the -users list, so we can continue the discussion there. Cheers, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center Anahi Ludue?a escribi?: > Hi, thanks Miguel. > I have another question: if I want to call the GoSub

Hi everyone, I was just wondering, does anyone managing production asterisk servers migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if there are some successful cases. Is your 1.6.0.X behaving well, with acceptable stability? Please share your experiences. Thanks, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57

Hi, I am using RT engine to log queue_log to a mysql database. My extconfig is [settings] queue_log => mysql,asterisk16_production Logging to mysql is working fine. But I find that the queue_log file now only has QUEUESTART lines for eg: 1250519094|NONE|NONE|NONE|QUEUESTART| 1250519186|NONE|NONE|NONE|QUEUESTART| How can I have queue_log in both db as well as in a file? thanks and

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Hi folks, I am struggling to install cdr_pgsql in asterisk 1.6.0.26. When I do the ./configure, it complains about the function PQescapeStringConn not existing in -lpq, so when I do a make menuconfig, I can't select the cdr_pgsql module. I am using CentOS 5.4 with the yum PGDG repository for 8.4 version. Some of the installed packages are postgresql-libs, compat-postgresql-libs,

Hi folks, I have a simple question regarding attended transfers. I have some queues where agents take calls and I have configured attended transfers between queues. That is, the agent dials the attended transfer extension that routes it to the aproppiate transfer queue where the second agent answers and they both talk for a while. Finally the transferrer leaves the call with *, connecting

I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL

Can i run asterisk on a openvz vps or do i need a kernel? I dont use dadi

Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the

Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:csibra at gmail.com

As Asterisk has inner problems and channels very often locks we have such script to restart Asterisk each midnight. We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions because there are too much changes which would brake our system (realtime/sip/iax2/cdr/etc/etc). Script soft hangups all alive channels in dirty way then kills Asterisk and starts it up. Hope

Hi, Is it me or is svn.asterisk.org down ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100413/d05b6b10/attachment.htm

Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not

Hi Guys, As it looks like from CLI command " show cdr mysql " , can somebody confirms that cdr-mysql creates persistent connection with in asterisk? " show cdr mysql " shows " connected to user at dbhost from 18 hours ......." -------------- next part -------------- An HTML attachment was scrubbed... URL:

After passing authentication, Then with this line, extent => 3616739999,5,DISA(no-password calls-outbound) As soon as the first digit of the intended number to be called is entered, the system does a Hungup 'DAHDI/1-1' It has done that no matter what I have tried. I am missing the boat somewhere. Anyone have tips? Cary Fitch

Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D

Hello, I have a var like ?blabla? with the ? I need to suppr the ? Is it possible with the ${var:x:y} ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this?

Hi all! I want to autopause my queue member when they are not answering within 20 seconds, and the autopause should affect all queues they are member of, not only the queue where the call was not answered. Is there a way to do that? The members gets dynamically added. I'm using asterisk 1.4.21.2.