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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

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 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 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

...What does "Extension" mean without >"Context" also being defined. I don't know what to set in my prefs.conf >file for extension... ?? >-Brett > > > > >>Message: 2 >>Date: Thu, 03 Jun 2004 09:27:44 -0700 >>From: Kyle Hagan <khagan@nuvoinc.com> >>Organization: Nuvo Technologies >>To: asterisk-users@lists.digium.com >>Subject: Re: [Asterisk-Users] Asterisk Receptionist manager program. >>Reply-To: asterisk-users@lists.digium.com >> >>I put a new version up last night. Caller ID shows up on the bu...

...meone mentioned there was a app_setDNIS function at some point but it seems to have > vanished again, or can i do it directly in asterisk/zaptel? > > Regards > > Darren Smith > Game Digital Ltd > > --__--__-- > > Message: 10 > From: "Kyle Hagan" <khagan@nuvoinc.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] CVS Problem? > Date: Wed, 30 Jul 2003 14:01:48 -0700 > Reply-To: asterisk-users@lists.digium.com > > This is a multi-part message in MIME format. > > ------=_NextPart_000_0050_01C356A3.1E84...

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.

...t of local patches and I was wondering if it were possible to get some of them merged into the Asterisk source base. Thanks -- David W. Chapman Jr. dwcjr@inethouston.net Raintree Network Services, Inc. <www.inethouston.net> --__--__-- Message: 10 From: "Kyle Hagan" <khagan@nuvoinc.com> To: <asterisk-users@lists.digium.com> Date: Thu, 8 Apr 2004 10:54:22 -0700 Subject: [Asterisk-Users] TDM Stater kit all working - WOOHOO - wondering about Asterisk FAX Support Reply-To: asterisk-users@lists.digium.com Finally got my Asterisk (Test) system up and working with a TDM S...

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

I know an email can be sent when a user get a voicemail message, but is there a way to send a message to a SIP phone to say they have a message? Or how hard would it be to write an app that could popup on a PC when there is a message in the mail box? Kyle

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

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 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'm having trouble getting the cards configured. I have followed the instructions but get errors. when I modprobe zaptel /lib/modules/2.4.21-0.13mdk/misc/zaptel.o: unresolved symbol proc_mkdir_Re6122de7 /lib/modules/2.4.21-0.13mdk/misc/zaptel.o: unresolved symbol skb_over_panic_R5ced0794 /lib/modules/2.4.21-0.13mdk/misc/zaptel.o: unresolved symbol devfs_generate_path_R4c75cb0f

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

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