similar to: Updated Patch to chan_agent.c for PREACKANNOUNCE

Hello, I think Device State for Agents don't work correctly My configuration: agents.conf [general] [agent](!) autologoff=15 ackcall=no acceptdtmf=# wrapuptime=5000 musiconhold=default recordagentcalls=no custom_beep=beep [2000](agent) fullname=Fulano [2001](agent) fullname=Zutano [2002](agent) fullname=Mengano queue.conf (Agents Related) member => Agent/2000 member =>

I have been successfully connected (incoming and outgoing) to FWD for a very long time. A few months ago, I changed from SIP-based FWD service to IAX2-based, and that went fine as well, both incoming and outgoing. At the time, I was running Asterisk 1.0.3 Stable. I rarely use the service, so other than noticing that I was always successfully registered to FWD, I didn't make or receive calls

This is weird, and could find nothing on the wiki or * google search. Regardless of the ackcall setting in agent.conf, if I have agents logged in to a phone on a zap channel, when a call is made the agent's phone rings, and when they answer, they have to press "#" in order to hear the announcement. If the same agent is logged onto a sip channel, then the announcement is played as

Following up on a thread that I started about Agents/Queue and acknowledging calls before bridging them... Greg Boehnlein said that he was putting his efforts into ICD. I downloaded and installed ICD, and I can get simple queue and agent stuff working fine, and see that this new design is much cleaner and more powerful. That said, in the sample conf files, the "acknowledge_call"

Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an "iax2 debug" output from a DIAX call to a local * server, dialing the extension that goes directly to the "demo" application. AsteriskHouse*CLI> iax2

I don't even know where to begin. I run a lot of production Asterisk servers, for a couple of years now, with no real problems. We built a brand new box, CentOS 4.2, and installed Asterisk 1.2.4 from source tarball(s). Built fine, and started up fine. Any attempts to do local audio (e.g. a "Playback(welcome)") results in complete silence. Worse, the Playback command will hang

I''d like to prepend TRACE_ to the macro names generated by dtrace -h. For example, change POSTGRESQL_LWLOCK_ACQUIRE(arg0, arg1) to TRACE_POSTGRESQL_LWLOCK_ACQUIRE(arg0, arg1) and still keep postgresql as the provider name. The reason for doing this is to make it clear that the macros are used for (D)tracing. In this particular case, without TRACE_, it appears like the macro is used

Hello, I upgraded to cvs-head over the weekend and now agents that are logged in on zap channels have to acknowledge ACD calls by pressing #, even though I have ackcall=no in agents.conf. This doesn't seem to be happening to agents on SIP phones, and it this is the first time I have had an agent logged in on a Zap channel since the upgrade. Is there a new config param that I am missing?

Can anyone recall a bug where asterisk crashes after a callback agent presses the ackcall key in 1.2? The last logged item was before safe-asterisk restarted was "delaying member connect for 2 seconds" We have seen it two times on a heavily loaded server (1.2.12.1), but cannot find anything documenting a bug or fix. Could be coincidence, but a google search showed at least

I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a

When using the callback feature on agents I notice that when the queue calls one of the agents and the agent picks up the call they hear nothing until pressing the # to accept the call. Only then does my announcement play back to the agent after which the call is immediately connected. Is there a way to have the announcement played to the agent before they press # to accept. I have ackcall=yes

I am trying to get remote extensions to work correctly with agents. I have ackcall=yes and have agents logged in to extension 101 using agentcallbacklogin with extension 101 defined as: exten => 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer)) This setup works great on local and/or voip channels, but on zap channels, the zap channel answers immediately as soon as it goes off

hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

I haven't seen anything this strange, and it's 100% reproducible. I'm hoping that there are some clever ideas out there for what to look for, since I can test to my heart's desire on this one... My Dad lives in Florida, and has a Bellsouth DSL line. Of course, he has a regular POTS line connected on the same line. He has the appropriate filters on every jack that has a phone

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Hey, We've been working on tracking all inbound calls to certain call-centre members and have hit a snag, it seems when a queue delivers a call to an Agent, chan_agent will call Local/whatever but provide no means of associating the call in the queue with the local call. To that end i added in an event called AgentAssociate which looks like this: 'DestChan' =>

Hi, I have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Yea, I have faked that with a silent mp3, but to do it right it should also be a config flag in the agent.conf file for each agent, prolly add another arg to each agent definition for the MOH class, & the arg 'none' means don't play music for that agent -----Original Message----- From: James Golovich <james@wwnet.net> To: asterisk-users@lists.digium.com