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I have a problem where if an agent's extension is busy and has voicemail the queue app will follow the dialplan and send the caller to an agents voicemail. This is really bad, because it takes the caller out of the queue when it hits that agent. But we also would like to have voicemail for some extensions like the shift managers etc. Is there s solution/workaround/patch? Thanks, -Ryan

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

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

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.

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

Here's the scenario, 5clients, each client has their own queue. There are 3 agents, and they're all logged into all of the queues. Using round robin or rrmemory an agent can get a call in queue 1, then hang up and immediatly get a call for queue 2,3,4,or 5 while the other 2 agents have not gotten a single call. How does one make it so that the round robin feature round robins to

For a while now in my call center I've been seeing calls that come in, hit an agent who is DND, and then bounce to the next agent, but instead of ringing for 20 seconds thing ring for about one ring and then go back to hold then cycle back through making it difficult to pick up the call. Then last night while testing a number that rings one sip extension and then another I noticed that

I've got a box running CVS-HEAD-02/18/05-02:27:22 with a digium single span T100P card. With increasing regularity we get inbound calls that will work fine for an arbitrary ammount of time (some calls last over 20min before the problem) then they are interupted by a short burst of static and then the line drops. On the console I see: == Spawn extension (did, 7214, 2) exited non-zero

Hey all, I thought I'd run this by the list before I attempt to go through official support channels (just because I never get anywhere). I've done 3 small business installations with asterisk and polycom sound point IP 500s. The first 2 installs run CVS-Head, the most recent install is just running stable. Now I have 28 polycom soundpoint IP 500's on their system, all using SIP no

I went to run my queue_log parser so that I could send out a monthly report to one of my customers, and I noticed that every valid call complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an ABANDON: Here is a complete-caller: 1123325015|1123325011.2|mainq|NONE|ENTERQUEUE||00110102102 1123325020|1123325011.2|mainq|Agent/21|CONNECT|5

So one of my employees just got married and is taking her husband's last name. As soon as her business cards run out she will be using the new last name for business as well. So for example is there a way to make it so that if someone goes to the directory to find her extension and they dial DOE it matches her or if they type SMI it matches her. Alternatively, can I make a new voicemail

I was wondering if anyone had any experience getting a 7960+7914 working with any of the chan_sccp modules. I've got a 7960G with 6.0(5.0) and a factory fresh 7960G with 3.1(MF.G2). I've got 2 7914s fresh out of the box brand new. I hook them up and all I get is red lights on all of the buttons. When I go into the phone to see what version firmware they have it says "Link State:

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

Hi I'm having trouble # transfering queue calls. in extensions.conf I have: [macro-queue] ; ; Places caller in queue ; ${ARG1} - Queue name to place caller in. ; ${ARG2} - Voicemail Extention ; ${ARG3} - Caller ID to Set. exten => s,1,DBget(temp=nm/on) ; Get Night key, if not existing,goto 102 exten => s,2,Playback(custom/500/10) exten => s,3,SetCallerID(${ARG3}) exten =>

Hello, I'm running Fedora Core 3 with udev, and asterisk/zaptel/libpri from cvs. I have followed the README.udev instructions replacing insmod with modprobe and rmmod with modprobe -r and adding the 60-zaptel.rules file, yet no matter what I do I still get this error on boot(or when I run service zaptel restart): [root@pbx ~]# /etc/init.d/zaptel restart Unloading zaptel hardware drivers: