[Asterisk-Users] Short burst of static then disconnect

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

Re: [Asterisk-Users] Forklift a 2000 phone PBX

cat3 will do 10bt, YMMV based on the install job and length of the runs, but for qaulifying the lines there is a simple one word answer, pentascanner. I've got one, if the job is in the sf bay area you can rent me for a day. -Ryan Michael Welter wrote: I'm staring at an RFP--this company wants

[Asterisk-Users] Queue app following dialplan

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

[Asterisk-Users] Dial timer problem? Short rings.

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

[Asterisk-Users] Problem with # Transfer from queue

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

Re: [Asterisk-Users] callback agents cannot transfer calls

I've got the same problem. It works fine on some of my older asterisk boxes that haven't been upgraded, CVS-HEAD-12/09/04, but not on the latest box, CVS-HEAD-01/19/05. I've tried both t, T, and tT no luck, I checked my features.conf and it has [featuremap] blindxfer = # i do not have

Re: [Asterisk-Users] callback agents cannot transfer calls

the # transfers... In older CVS from December 2004 the # transfer had to be terminated by # to start transferring. In actual CVS, you have to press # twice, type in the number, wait 2 seconds and the call get's transferred. Is this normal behaviour? Guido Hecken -Ursprüngliche Nachricht- Von: Ryan

Re: [Asterisk-Users] Calls directed via queue to unavailable device result in call acceptance

Well I fixed my setup by creating a seperate context for extension defenitions for agents. [agents] exten = 1000,1,Dial(SIP/1000,20,rt) exten = 1001,1,Dial(SIP/1001,20,rt) exten = 1002,1,Dial(SIP/1002,20,rt) and then had their regular menu accessable and inter office extensions in defualt

[Asterisk-Users] Balanced call distribution to agents logged into multiple queues.

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

[Asterisk-Users] weird ring behavior

In my queue I have about 4 agents answering at any given time, * has a tendency of rininging the first agent (rrmemory) for only half a ring then moving to the next agent, on the console it says it tried them for 20seconds. Anyone seen this or know where to look to fix it? Thanks, -Ryan

Re: [Asterisk-Users] ztcfg problems

30 line at the end of the start case in the zaptel init file. and then added an extra slink in /etc/rd3.d right before asterisk to run again. This corrected the problem. -Ryan Ryan Stark wrote: Hello, I'm running Fedora Core 3 with udev, and asterisk/zaptel/libpri from cvs. I have followed

Re: [Asterisk-Users] ztcfg problems

Michael Welter wrote: Ryan Stark wrote: 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

[Asterisk-Users] ztcfg problems

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

Re: [asterisk-users] Agent Callback Login in 1.4

I tried to use it back in 1.4.6 or so and it is horribly broken, I ended up rewriting the functionality with dynamic queue members in the dial plan. I really liked the call back agent feature set. I found it to be far superior to dynamic queue member alternative. -Ryan On 9/12/07, Anthony

[asterisk-users] T1 signaling configuration

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, EM (Robbed bit) Start

[Asterisk-Users] Queue_log all calls marked ABANDONED?

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

[Asterisk-Users] Voicemail/Directory, one person, one box, two last names

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

Re: [Asterisk-Users] Dead Polycom ip500

I've got an ip500 phone that had the same thing happen I've swapped power supplies, tried resets, reboots, leaving it unplugged for a week etc. The vendor I bought from turned out not to be polycom authorized even though the sales guys said they were and now I can't get polycom support on

[Asterisk-Users] Polycoms Go Silent after a a handful of calls.

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

[Asterisk-Users] Calling on all Polycom Experts

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

[Asterisk-Users] Polycom replacement handset

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

Re: [Asterisk-Users] Polycom Buddies in 1.6.6

So I've got a 601 (1.6.6) with the side car, and the buddy watch seems to be working but it updates the statuses unreliably. When I do a sip show subscriptions in asterisk it lists my phone 12 times and at the bottom it says 0 active SIP subscriptions(s) I've got an older CVS-HEAD build, pre 1.2,

[asterisk-users] Cisco 7914 with sccp

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