similar to: Queues

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122 at from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

Hi All, I'm running 1.6.0.17 and wanted to adjust the output from the command sip show channels. When there is a current call in progress the User/ANR field shows 10 numbers. The problem I'm having is that it is including a 1 such as 1949555121 which is truncating the last number. Is there a way to increase the number of characters displayed to 11 or to have the 1 dropped so I can see

Has anyone used the H option for Queue() with Callback queues? I want customers in my queues to be able to jump out to voicemail when they get tired of waiting, but in my setup when I pretend to be a customer and press '*' [when I am waiting in the queue] I see the message 'User hit * to disconnect call.' but then just jump out to the outer loop where queued callers wait to

Hi everybody, What I need to do is to run a context where I'll pass some phones (for example: 3 numbers). I need to make something like a followme, if the first phone is not answered, I'll call the second one, and so on. That dial plan is not the problem, my problem is when I execute the AMI, I'm using the Originate. It needs a channel as an argument, so the context can be executed;

Can anyone confirm that the feature listed below works? I'm using AgentCallbackLogin and it never seems to log the agent off if they don't answer. /etc/asterisk/agents.conf ; Define autologoff times if appropriate. This is how long ; the phone has to ring with no answer before the agent is ; automatically logged off (in seconds) ; autologoff=15 -- Go to

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

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello: ????? Suggestions for whomever maintains "R CMD": ??? ??????? 1.? Can you change it so it doesn't complain about the presence of ".travis.yml", at least on GitHub? ??? ??????? 2.? What do you suggest people do to find error messages in the output?? I ask, because I'm getting "build failing" from travis-ci.org, but I can't see what failed

Hello all, I'm going to tackle learning C this week, and start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Hi, Is there a way to detect if the caller will be entering an agentless queue? I'd like to be able to redirect any caller who tried to join a queue with no logged in agents, to be redirected to the groups voicemail. Is this possible? I know I could create a menu and an announcement for voicemail (should the user wish to drop from the queue); but they wouldn't know no one was taking

I need to map all of my NT id's to unix usernames. It looks like you can use NIS groups in the username map file. Why can't you put an NT group in there? Shouldn't I be able to map an entire NT group to a single unix id? Danny Travis danny.travis@exxonmobil.com

Greetings and Merry Christmas, We're trying to implements a queue and agents login mechanism on our Asterisk. After going over the documentation, we're unsure if we got it right. We wish to setup a "hotdesk" mechanism, where an agent comes to a station with a PC & IP phone (that is defined as a sip friend user in sip.conf), dials a certain number (agent login extension),

Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next

I've just installed asterisk as a new phone system for our office but am having difficulty with the queues. Specifically I need a way to redirect our sales queue to voicemail when no one is logged in to the queue. I see I can use the joinonempty=no setting, however this setting doesn't work if you use the agent functionality (at least not with AgentCallbackLogin). I could, of

I need clarification on couple of issues of Realtime Queue. It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk puts this Queue-Member relationship information into AstDB, So that on asterisk restart this can be preserved. My question is, why does asterisk not store call information for Queue (holdtime, talktime, W, C, A, SL%) in AstDB, So that it can also be retained

Hi Adam, I appreciate your feedback, and understand totally where you're coming from as far as the database perspective goes. For the first "draft" of the app, I think I'm going to let it default to using a conf file for two reasons. First, my setup currently does not utilize a database. I would like to move to that type of a setup in the future however. Secondly, seeing as

Hi I have a problem with the queues on Asterisk. The setup is Asterisk@Home v1.0 with Asterisk 1.0.7. I have 1 queue (4500) set up, with leastrecent strategy. There are no agents configured in this queue. Agents log in by dialing 4500* on their phones. All incoming calls are sent to the queue. Calls wait 120 seconds in the queue, and are then sent to voicemail extension 310. My problem is