similar to: Call Group

I seem to be encountering a problem with the db.h file when compiling gastman. Incorrect definitions for functions and so on. Here are the errors: If I don't copy the db.h from asterisk source to the include directory of gastman I get: gui.c:31:16: db.h: No such file or directory If I do then I get: gui.c:743: too many arguments to function This is for the line: if (!(res =

To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for

Is it possible to specify a 'wrap-up' time in a queue so agents will have a specified amount of time to complete tasks between calls unless they hit a key on the phone? As it is they can recieve a call moments after they hang up with no 'down time'. Thanks Jim Friedeck

Just cvs'ed about 40 minutes ago (10:15 CST 8-8-03). Segfaults when I use a queue app in many different scenarios. When calling phone is only member of queue I get a segfault. When 1st called extension is outside line I get a segfault. Many other scenarios as well. Unsure how to go about troubleshooting. Any ideas? Jim Friedeck

Thanks to Digium for the fine work they are doing on queue app! Does anyone know the progress on the monitor app recording to one file or synchronizing the two files' start and end times? Thanks. Jim Friedeck

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 =>

How do I stop asterisk when it is in a bad mood? It keeps dialing extensions and won't listen! I tried kill <PID>. No go. I don't want to have to reboot again. Thanks. Jim Friedeck P.S. I love it when my boss looks over my shoulder and I don't have an answer when he says: 'So, what are you doing?'

Dear all, I have some questions, I'm sure it's pretty stupid for most of you, but I need you guys to help me. Here are my questions: 1. Music On Hold, it doesn't play any sound on the parked call or hold call. But if I do ps-ax, it shows mpg123 .....( I forgot the exact line). I'm using slackware 9.1 2. I have fxs 3 port, and in my zapata.conf I have included callpickup=1-4,

Our CSR people need to be informed when a call is ringing in when they are on the phone. Is there a mechanism for informing an off-hook target channel of an incoming call? We have a guy who should get first shot at all incoming calls on our local lines and our customer service line. If he is on the phone, he should get beeped and then be able to place the current call on hold to answer the

I am looking for a way to quickly and easily test for on-hook channels from within a C-language AGI app. CHANNEL STATUS works but is a bit clumsy. I don't want to rely on strcmp'ing the returned '200 result=-1' as the meaning of this might change in the future. I am trying to create an ACD using MySQL and want to test each channel before I Dial it. Any ideas? Jim Friedeck

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

So I'm having trouble getting call-pickup working. Got a few different SIP phones (cisco 7940's and SPA-841s) all with pickupgroup=0 in sip.conf. I can't seem to get it working. This *is* possible from SIP phones, right? Do I need to add anything to my dial-plan? -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian Ridge Drive

Does anyone have use for a broadcast/annouce app? I wrote SystemAnnounce which will play a specified file to all active channels (in an UP or bridged state). This was originally to tell users to get off the system, but there are several other uses... I also wrote a new CallPickup and CallPark app, both of which work more as expected (supply actual extension numbers, etc). Let me know if there

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause

Hello All !!! I have 4 PSTN lines in the PBX server 1,2,3,4. First line will be used by only one extension (i.e. for the boss) for incoming and outgoing. This line is dedicated for him only.( The remaining lines will be shared by the employees 1) Group A have access to lines 2 , 3 & 4. 2) Group B have access to lines 3 & 4 3) Group C have access to line 4

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

I'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

I'm having a problem defining my channels Here is my zapata.conf ----------------------------------------------------------------------------------------------------- ; Zapata telephony interface ; ; Configuration file [channels] ; rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes