similar to: callers in queue passed to agents who accept only one call at a time

Hi, We?re looking at options for logging agents into the system programmatically via Perl/PHP and I was wondering if anyone else is doing this and if so, how. We're using AgentCallbackLogin now but would like to set up a web interface instead. I've been looking at Asterisk::Manager and didn't see anything relevant and wanted to ask the group before we dove into the Asterisk source.

Hi, I'm currently migrating to 1.4 and have problems changing deprecated AgentCallbackLogin to AddQueueMember. I have dynamic queues and dynamic agents (MySQL Realtime), and pseudo-dynamic agents.conf (with huge amount of possible agent numbers). Agent login is done trough manager API: * AgentCallbackLogin * QueueAdd In 1.4 seems AddQueueMember can do all the same, but there is no such

Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list What's the syntax of the QueueAdd command for the manager ? I want to add interface FOO to queue BAR using a manager. Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBWDAVJUm/Bor63CERAhbtAJ0YykdT+TNTNKXlC3NldtzCWLo1owCfZEoz

Greetings all, When using QueueAdd via the dialplan app, we are able to define an agent name... however, I don't see how this can be done via the Asterisk Manager. Am I missing something, or is this just not possible? Regards, Jeff

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a "reload app_queue.so" all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -------------- next part

Hello, Let say all the SIP devices will be registered on the proxy like kamailio. Agent is a member of Support and Billings Queues on the asterisk servers. Support queue on "Server A" and Billings Queue on "Server B" for example. This will be done via RealTime Queue. I want Agent to dial 1234 on a sip device and it will prompt to enter a pin number to Login via

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

I've been poking for the definition of QueueMemberStatus and all the source file indicates is that it is a integer member of the "member" structure. Anyone know where I can find the CONSTANTS definitions? -- Warm Regards, Lee

Hi all, As a disclaimer, I'm new to this list and very green administering CentOS. I run it on my test servers and do very basic networking and server administration (most users, permissions, and web app stuff). I have a backup server that I am using rsync to collect important in-process file for our digital library from both Windows and Linux servers, as well as a specific applications

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

I am running Samba 2:3.3.2 on a Kubuntu 9.04 Linux box. With the configuration file below the linux box can see and retrieve shared files on all 5 of my windows boxes. However, none of the windows boxes can see the linux shares as they are defined in the smb.conf below. The linux box shows up in the windows network neighborhood as Mercury1 but when I try to open it to show the shares I get the

Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via ->QueueAdd("4050", "Local/4053 at from-internal/n", 1) (not via agents.conf). SIP extension 4053 rings, answers and then decides to blind-transfer to ZAP/DAHDI extension 3666. The "show queue" command still displays 4053 as "In use". However, if 3210 calls 4050

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

I have a supermicro AOC-SAT2-MV8 and am having some issues getting drives to work. From what I can tell, my cables are to long to use with SATA2. I got some drives to work by jumpering them down to sata1, but other drives I can''t jumper without opening the case and voiding the drive warranty. Does anyone know if there is a system setting to drop it back to SATA1? I use zfs on a raid2 if

Hi all, Where can I find the status definitions for the status field below? Googled /web/voip-info.org and can't seem to find anything. Event: QueueMemberStatus Privilege: agent,all Queue: support Location: SIP/111 Membership: dynamic Penalty: 0 CallsTaken: 0 LastCall: 0 Status: 1 Paused: 0 Thank you! -- Warm Regards, Lee

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.