similar to: Voice mail notifications?

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

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

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.

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:

I have written an application (win32) to view parked calls. The reason i wrote the code is because my GS doesnt support the system reading the parked extension back to me after I park it. This will allow you to view where the call is parked, the caller id and when it was parked there. You can configure it to show parked calls parked by anyone or set it to a specific Device. If you put

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

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", "")

Ok its probabally something really eaisy im missing. I've searched the archives and voip-info. Asterisk is trying to send the email notification for voice mail. But the log says Invalid sender. Sender = asterisk@localhost.localdomain and not asterisk@somedomain.com as assigned in conf file. VM Config: [general] format=gsm|wav49|wav serveremail=asterisk@somedomain.com <----Actual

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

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

Thank you for all of the beta testing. New and improved graphics in this release along with drag and drop transfers and hold for all technologies. There's a screenshot on the link below. Also improved documentation so read the included README. There's also a sample xml configuration included. http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI Christian Hoffmeyer YottaDot

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]:

Does asterisk have a command to pickup another ringing extention? I've tried searching but couldnt didnt anything. Kyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040420/952ce2d4/attachment.htm

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

Is there any kind of Configuration guide available? I've been trying to get a SIP Soft phone to work and all I get is: NOTICE[81926]: File chan_sip.c Line 4716 handle_request): Registration from '<sip:xxx@xxx.xxx.xxx.xxx> failed for 'yyy.yyy.yyy.yyy' I know I'm missing something in the configuration. But don't know much about * yet. Thanks, Kyle --------------

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager status as dialing, and it just sits there with no tones, on the phone dialing out. If I dial,

We have been running into problems here, we have 2 PRI's when they fillup, All channels in use, and we dial more calls asterisk becomes unstable and crashes alot. We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by root@localhost on a i686 running Linux I know I need to upgrade. Is this a know issue?? Kyle

Hi all! I've got a Linux samba server and win 95/98 clients without problems. I've just installed 2 windows 2000 clients and I have a problem with one of them. One of w2000 works fine, and it logs on samba domain, but the other it doesn't log on. The account of this machine was created , but when I attemp to log on, windows 2000 says to me "Cannot join domain, the credentials