similar to: Having a Blonde moment.

http://bugs.digium.com/bug_view_page.php?bug_id=0001693 This patch adds a lot of options for AgentLogin/AgentCallbackLogin Please test and respond in the bug tracker! /O ------------------------------------------------------------------------------------- "This patch adds quite a few new features into __login_exec () of channels/chan_agent.c for AgentLogin() and AgentCallbackLogin(). Only

Hi All, I am trying to implement ACD using Asterisk 1.2.18 and I've chosen AgentCallbackLogin for login purpose. One AGI is written which will actually get executed when agent dials '1001' (say) from his SIP phone and enters into the queue. Second AGI gets executed when the Dial operation is performed. I see the agi_uniqueid obtained from both AGI instances are different and I

Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/<number>@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc).

Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/a4f29a97/attachment.htm

Hello all, I am building a software based on asterisk to handle incoming answering service calls. I have one problem that I have not been able to figure out a reasonably priced solution to: The goal of this software is to allow the agent to be able to do their entire job from the desktop. The only thing that seems to be a problem is getting the operator (agents) headset logged on to the

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the

I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11,

I am inquiring about setup of the SWAT utility I have installed Red Hat 7.0 with samba installed during the initial setup of Redhat. I have two network cards installed in my Server and I am connected to the internet via a Cable Modem. When I try to start SWAT netscape displays the message that it cannot find the local host on port 90. I have downloaded the book over samba and I have also tried to

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set

Can someone tell me why this is not valid... [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@Local) exten => 2000,1,Macro(DialProxy,115551212) exten => 3000,1,Queue(testq||||45) while this is: [start] exten => 1000,1,Answer exten => 1000,2,Wait,1 exten => 1000,3,AgentcallbackLogin(1000||2000@start) exten =>

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>

I am developing an ACD front end using Asterisk 1.2.14. I heard that AgentCallBackLogin will be deprecated in future version of *. Is this true? If it is, how can I use AddQueueMember to replace AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have multiple queues and a lot of agents defined in queues.conf and agents.conf. Each agent may login more than one queue. It

I need to set up the sound card of a server to use in an overhead paging system, as normal I am testing this on my home machine first (which has slightly different Hardware). I'm using chan_alsa with the Intel HD Audio driver on an Intel 82801G (ICH7 Family) sound card. I am running Asterisk 1.4.17 and have a fully loaded TDM400P as a timing source. When calling console/dsp (using

Hi List, Ive got a few * boxes connecting together, one box is doing AgentCallBackLogin() and then the 2nd box is holding some phones at a remote site. I have users login to the main box and * shows the user is logged into a extension that resides on the other box, problem is, when I go to make a call to a agent, I get "May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:

Hi, I am a beginner who began to use Asterisk in this July. Last week, I made two addons for PostgreSQL (cdr_addon_postgresql.c and res_config_postgresql.c), because I use not usual MySQL but PostgreSQL. # Of course, not scratch build but modified version 1.4.8. But I don't know how to change configure script, menuselect-xxx, ... etc, and how to merge my sources to addons tar-balls. Anyone

Hi, So first daft question with Centos 6 (someone had to be first!) I've setup Centos 6 as a Server but as with Centos 5 it used to boot into the GUI but v6 doesn't do this, startx etc doesn't seem to work to launch the GUI Any suggestions on how I can get this to work? Thank you Keith