similar to: Re: Ring requested on channel already in use

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

Hello, We are running dovecot 1.0.5 on a test server, with FreeBSD 6.2 (though I have noticed the same problem since dovecot versions in the 0.99 range). We don't have very many simultaneous pop/imap users, but we have a proliferation of pop3-login processes. Currently we have 128 such processes. We have 11 imap-login processes, but only a few actual imap processes running. Is this normal?

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!

Hello, We're having a problem in Dovecot 1.1.8 with a failed assertion on certain mbox format mailboxes. It happens both with deliver when it attempts to delier to the mailbox, and with IMAP connections for the affected box (though I'm not sure what they're doing at the time). Mar 3 12:55:26 <snip> dovecot: Panic: IMAP(<snip>): file istream.c: line 81 (i_stream_read):

Our current virtual mailbox configuration is not compatible with one of the assertions in the dovecot quota plugin's assertions in quota-dirsize.c. I believe the assertion is incorrect, but I would also be happy if I could get the same result with a better configuration setting. Here is a sample passdb entry which causes the quota assertion to fail: test at

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Hello. I have a likely unusual request regarding IP based virtual dovecot users. When you specify a passdb passwd-file name containing "%d", then the domain portion is stripped from the login username, before the user is checked in the passwd-file. However, if you specify a passwd-file name containing "%l" (the local IP), the domain portion of the login is not stripped off

I just experienced something I'd rather not experience again. Using a SPA-841 SIP phone connected to our Asterisk server, someone dialed their own extension, answered, and then transferred the call using the phone's "XFER" soft key. This does a SIP REFER. Now, the phone has dropped out of the loop, and Asterisk has connected the two call legs into a loop, as far as I can tell.

Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. One feature the SPA-841 has, which I can't figure out how to implement on the snom 190, is the "make/accept calls without registration" feature. Or more specifically, "produce a dial tone even if I'm not registered." I would like to set our

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

Hello, In our configuration, we are using a "passdb passwd-file", with "user=" directives in each username, and a separate "userdb passwd-file" which contains the target usernames for the "user=" directives. This works fine, for normal logins via POP and IMAP. For customer support testing purposes, we also set up a temporary "master=yes"

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

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

Hello. A while back, I noticed an odd problem with our SPA-841 phones connected to Asterisk. Now we are having a different odd problem, and I'm not sure if they're related. I wonder if anyone else has experienced anything else like this, and/or if there is any reasonable explanation? Occasionally, one of our SPA-841's will spontaneously start up with "Welcome to Comedian

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 =>

Russell Bryant wrote: > Greetings! > > Version 1.0.8 has been released of Asterisk, Asterisk-addons, Zaptel, > and Libpri. This release contains a significant amount of bug fixes > (possibly the most of the 1.0.X releases). Tarballs are available on > the asterisk web site as well as the asterisk ftp server. Thanks! I appreciate the effort you put into these releases.

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, 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:

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: