similar to: Dialplan SIP call back problem

Hello, I am using Asterisk 1.4 version. How to dial multiple numbers per second through asterisk manager???? Thanks and regards

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

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

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 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

I am trying to find out how I can tell the length of a string actually CALLERID(num) in the dialplan. How is that done? If need to test the length of the CALLERID(num) if its less the 10 digits I need to set it to a known value or insert 0's at the beginning until it is 10 digits in length. My PRI provider needs it set to 10 digits always. Thanks, Jerry

I've been using Asterisk for a few weeks now. I have a (1) BT100 phone and a Sipura-2000 for all my analog phones. All has worked rather flawlessly, until today. I was on the BT100 phone today. During my phone conversation, the BT100 disconnected and went into a "click" mode. 2 "clicks" per second I think. Asterisk was fine, I picked up one of the analog phones,

When using call files I see Dial command when on the CLI however when using the AMI I dont see the originate Dial. Is there a logging level that will show the originate dial on the CLI? If there isnt currently a log state for that shouldnt it do that just like the call files do? My logger.conf has: ; debug => debug console => notice,warning,error,dtmf ;console =>

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: #

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

Date: Fri, 1 Oct 2010 18:40:40 -0300 From: Rodrigo Lang <rodrigoferreiralang at gmail.com> Subject: Re: [asterisk-users] AMI Originate To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <AANLkTikV+32vKVSkAFmkDciOPn+rO=k3jYJmsZLNj1QS at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 3

I know I'm doing something simple and wrong, but I can't quite figure it out: Example (executing system command): Action: Originate Channel: Local/1 at dummy Application: System <http://www.voip-info.org/wiki/view/Asterisk+cmd+System> Data: /path/to/script I keep getting a "Unable to request channel" and am not sure what it is looking for in place of "Local/1 at

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

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

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

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

Hi All, I'm trying to configure 2 GS BT100 connected to asterisk and Welltech 2 ports FXO gateway. I configure WellTech 2ports FXO and GS BT100, both GS BT100 can call each other without any problem but when I tried to call a local extensions connected to my Welltech FXO gateway, I couldn't hear any voice on both ends. I would like to ask if anyone has ever encountered this kind of

Hello! I?ve an issue whit CDR using asterisk 1.4.23.1. I?ve configured mysql to store cdr information, but, while I put into cdr_mysql.conf the field ?userfield=1? and doing a query I found that this field is empty in the cdr table. On the other hand I can?t find records in the cdr table that show me calls generated through AMI using Originate Action, that?s calls are not stored in the CDR, but I

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,