similar to: Parked Call Viewer Application - Freeware

We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

What is the state of speex support in asterisk? I saw the codec seems to be there. Can speex be used on IAX2 links? Is there much work still to be done? many thanks, --J.

I have two contexts and there I have some sip clients and some iax clients, in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax clients I have some extentions like 2000, 2001, 2002, 2003, etc. My extention is 2003 when I make a call the manager program show me that the extention 20 was originating the call (in red, and show my name/extention)! When I make a call to

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Is there a DS# (T3) card that will work with asterisk? OR a card that supports more that 4 T1's per card? Kyle

Has anyone managed to set up a moh server for Asterisk? Reason would be to offload processing off the asterisk box, onto another system. The wiki is a bit light on details. If anyone managed to get it up and working, what software did you use on the server side, and what client app did you use? Mpg123? Mpg321? Madplayer? Something else? Also, putting legal ramifications aside, it'd be nifty

Anyone having trouble parking calls? I haven't tried it in a while, but it seems to have stopped working. If I dial 700, I get a invalid extension. I have "include => parkedcalls" in the correct context, and I can dial 701, which tells me no call is parked there. Any ideas? Parking.conf is stock.

I've searched WIKI and Archives but nothing. Im getting: -- Called username:password@iaxtel.com/1800somenumber@iaxtel Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answer at this time -- Executing Hangup("SIP/104-b8eb", "")

Is there a was to reset asterisk by dialing? Eg *99 and have it execute the commands to stop now and restart? Also eg *56 to reload? The reason for the restart command is over the weekend it started acting weird, the voice got really choppy and aver i restarted asterisk it was fine. We are just in testing right now but it would be nice to know how to do this if it were to happen again and im

We have install a DS3 with 28 DS1's we have an Adtran MUX breaking out the DS1's, we are trying to setup the system with 2 dchannels for each 4 DS1's. Everything looks fine when modprobe zaptel and wct4xxp and ztcfg -vvvvvv but when I asterisk asterisk it says: Aug 10 16:33:32 ERROR[8954]: chan_zap.c:6750 mkintf: Channel 24 is reserved for D-channel. Aug 10 16:33:32 ERROR[8954]:

Hey All, I am working on a SIP Call bridging application. Every time I receive a incoming call in Asteriskserver1 my AGI should alert to AsteriskServer2 and AsteriskServer2 should callback to AsteriskServer1 and call should be bridged on specified extension. (making call in this way is customer requirement) Every time I receive a call in AsteriskServer1, I Park the call and through AGI,

So, I can't figure out how to get the parkandannounce application to work the way I want it to... I have cisco 7960 IP phones using SIP, and this is what I have in my extensions.conf: exten => 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1) exten => 700,2,Hangup and in my parking.conf: [general] parkext => 700 ; What ext.

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

I've tried to find other threads with this same topic, but haven't found any... Apologies if this already being discussed.... Running asterisk 1.4.13 (upgraded from 1.4.9) and zaptel 1.4.4. Having an issue with (I think) parked calls. We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a

Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered

Hi all, how can I catch the event generated when a parked call is hung up? In my dialplan, when arrives a call to a specific number, Asterisk parks the call and announces the parking slot to a number. But if the user hangs up the parked call, I don't know how to catch the event, from dialplan, that removes the call from the parking slot. I want to know if there is a method for do this.

I know the call parking feature changed in 1.4.23.1 to fix some serious issues. I'm seeing a major change though which I find disturbing. A person parks a call by transferring it to the parking position (700). When the timeout value is reached, the call is NOT returned to that device, but rather the 's' extension of the phone's registered context (in this case [unrestricted]).