similar to: Deadlock detected in asterisk-1.8.9.0 x86_64

I've been running into an issue where chan_agent gets stuck and all queues stop working. Here's a show channels from when it's stuck: Channel Location State Application(Data) SIP/56-be24 s@macro-stdexten:10 Ring Dial(Agent/19|50|tw) Local/*14@agentlogin *14@agentloginoff:1 Up AgentCallbackLogin() Local/*14@agentlogin *14@agentloginoff:1

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

In my application, I am using AMI to run an Originate command between a channel and a dialplan application (NOT a context). In my case, the application I want to invoke is FastAGI. The Originate AMI command works correctly, but Asterisk generates a very short (0-1s) duration for the CDR that results from this call, regardless of the time spent running the FastAGI application. I want the CDR

Anyone know a good replacement for phpagi? Unfortunately development stalled long ago and it has not been updated. What is the best solution for AMI and AGI on PHP? Thanks. -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24

Which kind of transfer do you use? Try using the # transfer. Hope that helps.. Guido Hecken -----Urspr?ngliche Nachricht----- Von: Diego Magalh?es [mailto:diego@redetaho.com.br] Gesendet: Mittwoch, 2. Februar 2005 17:21 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] AgentLogin / AgentCallbackLogin transfer problem Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and

Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and having a problem with call transfers using the cmds AgentCallBackLogin and AgentLogin First Case (using cmd AgentCallbacklogin): When the incoming call comes and enters the queue, the agent logged in answer the call. But when I try to transfer this call to another agent, the incoming call is dropped. I don?t receive any error

In order solve my incoming caller ID problem, I upgrade the dahdi to version 2.6.1 from version 2.4.x. After upgrade, I found the echo cancellation doesn't working (I'm using Digium AEX800B PCI Express card). I can hear my self talking on the phone. How to solve this? I think I need to recompile dahdi 2.6.1 with OSLEC support? how? [root at callcenter ~]# dahdi_cfg -vvv DAHDI Tools

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

and I found a reference to an AgentLogin.rtf. Looks great, except I can't get it to work. queues.conf: [sjs-testq] music = default timeout = 1 retry = 1 maxlen = 0 member => Agent/10001 agents.conf: agent => 10001,1234,Steve Sobol extensions.conf: (I have a phone line set up on which the main menu tells you to press 1 to be added to queue. Pressing 1 lands you here) exten =>

HI all: Iam planning to use my asterisk box as callcenter?,any one can advice me with the best callcenter open source program based on asterisk . ? Any help will be apreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100307/116f1b75/attachment.htm

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: #

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

The Asterisk Development Team is pleased to announce the release of Asterisk 1.8.9.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

I have 4 sipuras 3000 in a small callcenter connected to the PSTN receiving calls and forwarding them to Asterisk and viceversa. In addtiion I have some x100s, linksys FXSs, etc Strange things are happening with the Sipura and Asterisk which I cannot seem to figure out. During off hours at the callcenter, when no one is placing calls, if I place or receive a call with any of the Sipura,

Hi, Can some one tell me what is the meaning of all the fields of show queue callcenter? for example in my system it gives: callcenter has 0 calls (max unlimited) in 'roundrobin' strategy (33s holdtime), C:429, A:12, SL:0.0% within 0s How is the holdtime calculated? what is A and SL? Also how can I see which of my zap interfaces are busy currently? I did a zap show channels I get

Followed; http://www.voip-info.org/wiki-Asterisk+Agents agents.conf [agents] agent => 1001,4321,Ben Dover queues.conf [queue1] member => Agent/1001 extensions.conf exten => 28,1,AgentLogin(1001) exten => 29,1,Queue(queue1) But when I call number 28, I get: "Please enter your password followed by the pound key".. but when I enter the the password, 4321,