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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

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.

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All, I have a question about calculating the value for the -E stride option to mke2fs. The mke2fs man page says stride=stripe-size Configure the filesystem for a RAID array with stripe-size filesystem blocks per stripe. So stride = size of stripe/blocksize. The size of a stripe is the RAID chunk size * the number of drives in the RAID. My question: are parity disks

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

All, I'm requesting comments from the expert readers of ext3-users on these notes for tuning ext3 for performance. Most helpful would be feedback pertinent to RHEL 5; as XFS isn't supported under Red Hat Enterprise Linux these items are an attempt to match XFS performance with ext3. These items were culled from a number of sources. Will they be effective for achieving the performance

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

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

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]:

Hello, We are looking to adjust the font size of the axis annotation on the graph that results from use of the metaplot() function. Metaplot seems to respond to cex and cex.lab to change those graphical parameters, but it doesn't respond to cex.axis. Is there a way to work around this by creating a customized x-axis, and if so, how? Thanks for all your help. Syntax is below. Best, Dawn

Currently I have a setup that involves connections from several different providers ranging from 6 mbit DSL''s to 10 mbit fractional DS3 connections tied together to provide multipath high availability internet in our area for schools and businesses. We recently extended our coverage to another area which has several other ISP''s. The box that is doing multiwan right now is the

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

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

(Perdón, no puse asunto en el post original) Ah, corremos R 3.0.2 y l alibrería MASS 7.3-29 Estimados todos: Os escribo porque tengo un problema que nos está dejando un poco trastornadillos. Probablemente sea uno de esos que los ve un experto a un kilómetro y dice ¡oh! toca aquí. Pero nos trae de cabeza. El caso es que aquí usamos un script para calcular ciertos parámetros relacionados con el

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", "")

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