similar to: Queue app following dialplan

Hi, my situation is: incoming call goes into the queue and is picked up by callback agent. The agent then wants to transfer the call to another device (another SIP phone). But 'transfer' button doesn't work and '#' button attempts to start channel monitor. Tried with both Queue(testq) and Queue(testq,tT). Is it meant as a feature that agents won't transfer calls at

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

I'd like for our custom soft phone to be able to know what queue, and/or what DID is calling an Agent's phone before the agent picks up. The agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple queues so it would be nice if they could get a pop up window telling them who's on the line before they pick up and hear the announcement telling them that. I'd like

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

http://host-a.starnetworks.us/Members/kpfleming/spring_von/photoalbum_view Enjoy!

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

Hello everybody, I have started using Asterisk in a call center with ACD. I have noticed something and I wonder if anyone knows whether it is a bug or a feature! I am using Queue application to ring a number of agents that have logged on using AgentCallbackLogin. Now, while an agent receives a call from the Queue they cannot logoff using AgentCallbackLogin. Instead the Agent is asked for

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'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

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, we are looking for an opensource or commercial * based Call Center. Full ACD, call monitoring, multiple queue, IVR, voicemail, management, reporting, CDR, etc is needed. over 100 seat can be the initial target and will grow in a very short time. SIP phones will be used and multiple E1 lines incoming, so to provide full failover a cluster of * machines or some other form of redundancy

I am running Asterisk 1.8.10.1. I am trying to set up some routing in my dial plans and having some issues (the issue being that I don't quite understand all of the syntax and patterns that can be used: Examples: DID1 = 6140000000 DID2 = 6140000001 CNAME1 = 6149999999 CNAME2 = 6149999998 CNAME3 = 6149999997 context1 context2 context3 I have looked at several examples (patterns) and I

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

Hi all, I ve setup a queue with 2+ agents for managing our inbound calls from customer. Using Asterisk 1.2.18 in a CentOS box. Agents login using AgentCallbackLogin application and I use a BASH AGI to accomplish this as there are some validations done with MySQL DB. Im aware that transfer could be done with option 't' in the queue() application and I was able to successfully transfer

Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives -> put it in a queue -> remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone

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 =>

This is weird, and could find nothing on the wiki or * google search. Regardless of the ackcall setting in agent.conf, if I have agents logged in to a phone on a zap channel, when a call is made the agent's phone rings, and when they answer, they have to press "#" in order to hear the announcement. If the same agent is logged onto a sip channel, then the announcement is played as

Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is "busy". Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks

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