similar to: ACD with polycom ip phones

Displaying 20 results from an estimated 3000 matches similar to: "ACD with polycom ip phones"

2006 Feb 23
2
Polycom 501 ACDlogin
Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function
2006 Feb 23
2
SV: Polycom 501 ACDlogin
Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com
2006 Jun 24
2
Asterisk ACD with Polycom IP501
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
2006 Jan 05
2
Call Group Limit
I recollect that there used to be a fixed, finite limit to the number of call groups that could exist. Does anyone know if that limitation still exists in 1.2.1, or maybe where I could look in the code to find out if it's a fixed length array or not? Thanks. Doug.
2006 Mar 25
2
Copying SIP Subscriptions
I'm pretty sure I already know the answer to this, but... Is there a way to copy/transfer/replicate sip subscriptions from one asterisk system to another, for the purposes of HA? You coudln't even write a script to do it I don't think. You can do an 'asterisk -rx sip show subscriptions' but there'd be no way to repopulate it on a second system. Yes/No? Doug.
2006 Feb 28
3
Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?
Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia
2006 Jun 13
1
Polycom Queues
Has anyone integrated Asterisk Queues with Polycom phones? What I'd like to do is display the agent status next to their appearance. I don't see much discussion about this. This is not the same thing as setting <bw>1</bw> against the appearance in the phone directory. Thanks Doug.
2006 Jan 14
3
1.2.1 "Silence suppression is disabled" what the hell?
I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence
2006 Jun 12
2
Hitting * in a queue call hangs up?
Can anyone explain why when I hit * (as in *2 to transfer) a call that has come to me in a queue asterisk disconnects the call? All I have to do is hit "*" and the call drops.
2007 Jan 15
2
Queue cmd option 'i'
Using Asterisk 1.4, on the console 'show application queue' mentions an option 'i' that should "ignore call forward requests from queue members and do nothing when they are requested." Does this work? My assumption is that the member whose next according to the queue strategy should get the call even if they have forwarding enabled on their SIP device. The forwarding
2006 Mar 27
1
FW: Re: Fw: anybody has SIP realtime working ?
Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000
2008 Jan 04
2
Agents and AddQueueMember
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
2006 Jun 15
1
Distributed ACD Queues
It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle. dundi.conf: 180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial 180q =>
2006 Jan 30
3
How many digium cards per server ?
Hello, How many digium cards is supported per asterisk server ? 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
2006 Jun 12
7
Can this config sustain 30 users?
I have this server I need to put to work. The option I have is to make it work as a small office PBX with SIP users and a Digium E1 Card for PSTN service. 24 SIP users and one E1 card in an Intel 945board (533 Front side bus) with 1GB DDR 533mhz of ram, one Pentium Dual Core 2.66 ghz (FSB 533MHZ) and two 80GB SATA disks. Can the box sustain the load? I can add another 1gb of ram if necessary.
2014 Oct 22
1
[asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang)
On Oct 22, 2014, at 11:47 AM, BJ Weschke <bweschke at btwtech.com> wrote: > On 10/22/14, 12:14 PM, Paul Albrecht wrote: >> On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: >> >>> Paul Albrecht wrote: >>>> Really? Shouldn?t something this major affecting the entire Asterisk >>>> community get discussed on the lists?
2005 Oct 15
6
ACD calls to busy agents
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
2005 Sep 27
2
Integration with NMS AG-E1/T1
I want to replace a custom PBX, that is infront on a IVR system based on OLD NMS AG-E1 Card. The Cards is configurated with CAS Digitalmode, someone can give me some info about Digim Cards CAS configuration i need a conversion Table? I wanto to don't touch configuration on winbox, i want only replace HWPBX box with asterisk. Diagram Telco E1 ===>Proprietary PBX========(CAS)===>IVR
2006 Apr 07
2
407 proxy authentication
Hello, Asterisk sent back 407 proxy authentication . How can avoid this ? I set insecure=very without success in sip.conf and my sql server . 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
2006 Jan 12
4
dCAp
HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk?s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and