similar to: ACD queue question

I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working.

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

I don't know if this is possible, so I will let the collective decide. Here is what I would like to do. BossA calls BossB, BossB's admin assistant sees the call from BossA on her phone. CallerID would look something like: BossA to BossB : on her phone. And she would be able to pick if BossB was not in his office. I am sure this is possible, but I do not know where to start, or even

I have spandsp installed and working, but when it emails using Scotts mailfax, the attachment is a dat file. I tried to rename the file to .tiff or .pdf, but it will not open. In the /var/spool/asterisk/fax folder, that faxes are there as tiffs, and I can open those without any trouble. The problem is in the conversion from tiff to pdf. Is there another package that needs to be installed for

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

Good Day again list, Encountered another problem in the ACD queue... If I use the ADDQueueMember to dynamically add members as foolows, exten => 403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204

I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the music on hold does not work. More Detail: While I was running asterisk 1.0.5, when someone called into an Polycom IP500 and was put on hold via the Polycom "Hold" button, the hold music would play. After upgrading to 1.0.6 that does not work. But if I set up an extension to play the hold music, it plays.

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk??

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

Hi, I am doing some tests with Asterisk's ACD capability, and as far as I could go I have realized that each agent defined in agents.conf must keep a session (call) open with Asterisk in order to be considered online. When a user calls, the agent receives a beep notification in his softphone and he answers to the pending call in the open channel and after the call ends he remains on the open

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

I have a couple of real quick questions for ya'll. I am trying to setup * in a test scenario to emulate my company's 20 agent ACD. 1) First, I am curious if anyone has any configs that they can throw up on the wiki or just send via e-mail. I have a bit of it pieced together and am able to make some very basic queued calls, but I am sure that there are much smarter people out there

Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part..

Hello, I have three queues (F1,F2,F3) with default queue weight and three agents (A1,A2,A2) with default agent penalty. If the three agents are busy and tt same time a caller (C1) enter in the queue F1, and after 20 seconds a second caller (C2) enter in the queue F2. So, few seconds later, the agent (A1) state comes to availabe. In this case the asterisk deliveries the caller (C2) to agent (A1),

Any advice on a good conference phone that works with Asterisk? I like the Cisco line and was wondering if anyone has used the 7935 or 7936 phones. From what I can tell they don't have a sip load. Has anyone verified this or gotten an ETA from Cisco? Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the