similar to: Extention pickup

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 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

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

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

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

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

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]:

I have created a rxml template and I am sucessfully generating valid xml. However, the file does not have an extention on it. This is creating a problem for one of the applications running on my computer, when referencing the file off the server it requires the .xml file extention. Despite being vaild xml it needs that extention. I am lost at how to solve this problem. I am still working in a

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Buenos días, Como resultado final de unos análisis espaciales que hago, genero unos mapas y gráficos. He creado una ficha como esta a mano: http://dl.dropbox.com/u/7026890/CM/RSesmaf.html en la que presento un mapa resultado con los puntos de muestreo y una serie de gráficos de evolución. ¿Sería posible generar algo similar con R? Por ahora lo que he encontrado sería escribir el html línea a

Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error ---------------------- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' --------------------- Can you tell me what

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

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 --------------

Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -------------- next part