similar to: Agentcallbackogin without any user input after extension is dialed.

Hello: I want to have a local telephone number that, when the people calls this number (via mobile or normal PSTN), the voip provider stablishes a SIP session to my asterisk box. It is possible? If yes... What providers have this service in Europe? It is difficult to configure and get things working ok? Will my asterisk box see the mobile or normal PSTN phone# that is calling the number

Hello everyone! I 've looking for carriers which can terminate my international calls. They must accept payments from Argentina and give me interconection to my Asterisk. I'd appreciate your help or recomendations. Regards. -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Share your knowledge, use free software.

I'm developing a small linux/speex-based audio streaming (VoIP-like) application, and am looking for a binary windows app that can play speex content (at least from a file, preferably from an IP data stream) out through the windows sound system. This would be very helpful as a test tool. Any recommendations? I'd prefer a binary as I don't have development environment available for

Dear all, I have been using Speex during last year over PC with very good results. Now I'm trying to move my code to the DSP TI C6713. It's the first time I work with a DSP and with Code Composer. I use speex quite well but I'm not very good at compiling for different platforms. After reading all manuals I tried to compile the speex library with Code Composer and the speex options for

Interested in building a speex codec (basically audio <-> speex <-> data stream) using TI's small MSP430 microcontroller. Is there any way to estimate feasibility based on CPU requirements? Example - speex is happily encoding on an old Pentium-1 processor (166 MHz) using about half the CPU (as reported under Linux); the TI microcontrollers are much slower yet (8-16-25 MHz) and

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

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, My queue used to work fine until I upgraded to 1.6. I am getting the message: No application 'AgentCallBackLogin' for extension (default, 31001, 1) After some rearch I learnt that AgentCallBackLogin is removed in 1.6. Any one has a configuration that works in place of AgentCallBackLogin in 1.6. -- ond -------------- next part -------------- An HTML attachment was scrubbed... URL:

I would like to know if AgentCallbackLogin will be discontinued anytime shortly in Asterisk 1.4.x . I've read a page in voip-info that said so [1], but it was not an official announcement. I have to be sure, because I'm in the process of setting up a medium-to-big callcenter with Asterisk and calling the agents when a call is placed in a queue is a requisite (I've also read it can be

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes

Anyone know if - it is possible to limit 1 agent per extension where the last agent to log in overrides any previous agents or - a Command/application to clear all agents logged in on extension Does this look like it would require a custom mod to do it? J __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out!

I am using AgentCallbacklogin to logon agents. I am trying to avoid agents being logged in more than once in different extensions (is this a bug?) by passing the callerid to the AgentCallbacklogin funtcion as an option. The problem is that by doing this, agents are not asked for an extension and they cannot logoff (by pressing the #). Any ideas how can agents logoff? -------------- next part

Hi, I am on my next venture now, Need to set up 3 queues. I would like these setup using the agentcallbacklogin. Does anyone have an example of this? I have looked through the wiki , but all that did was confuse me. One of the problems I'm having is how do I configure my extensions.conf to dial the agentcallbacklogin -------------- next part -------------- An HTML attachment was

Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I'm using the apllication AgentCallBackLogin so agents can login to a queue. They just need to enter the password, the CallBack Extensions is the ${CALLERIDNUM} Is there a way to AgentsLogOff withou using the AgentCallBackLogin application. I don't want the user to enter they CALLERIDNUM. Regards -----BEGIN PGP SIGNATURE-----

Hi! I'm trying to transfer a incomming call from a Queue to another extension. I'm receiving a call from a queue with the AgentCallbackLogin. The queu is as following: Queue(sales|t) Which should allow transfers. So as soon as the call is answered I would like to be able to transfer it When the agent presses the # I get the dialtone but as soon as I press any digit Asterisk tells me

Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano