search for: radioshack

...the call hangus up. Also, for example, if a call is done, and an other extension hangup, there are some probability that the other extension hangup, kill my call.. I have any clue about this. I?ve watching the * console all over the day, and no clue, neither... logs, says nothing.. I?m using radioshack adsi 2.4ghz phones. TIA

Anyone using this 110-220v 9v 1500ma supply? If so, which DC plug adapter does the IAXy need? A friend brought the wall wart over here for me but she said you get one plug free and she didn't know which one it took. Does anyone here know? http://www.radioshack.com/category.asp?catalog%5Fname=CTLG&category%5Fname=CTLG%5F009%5F001%5F001%5F003&Page=1

...for sufficient time, but still didn't make a difference. I have tried resetting to factory default - no luck. I have tried switching the power outlets on the wall, no difference. I realize this isn't really asterisk related but it's just damn weird, anyone have any ideas before I beg radioshack for a return? I've only had the phone for about 3 months so don't see how it could die already. TIA, Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions PSTN: 1.877.999.4678 ex. 6400 FWD: 472645 IAXTEL: 1.700.761.1828

I have a standard PS1 controller hooked up via a radioshack PSX to USB converter. For some reason, only the Up and Left buttons on the D-Pad work. Here's the error output. err:dinput:JoystickAImpl_Unacquire Failed to set autocenter to ffff: 9 Bad file descriptor

Hi! I am looking to calculate the end-to-end delay between two soft phone/hard phone. I have asterisk server and configured ntp server on the same machine and synchronized it with ntp pool. I have seen that Wireshark can be used to check the jitter. But I am not sure how can i calculate the end to end. May be this is not related to the mailing list topic but please help me if anyone has some

Ok, this is probably a stupid question, but I can't seem to find anywhere where it tells how to put someone on hold on a zap channel. Flash gives me a dialtone and # tells me to enter a new extension, how can i just put the caller on hold. Pressing # then hanging up drops the call. Is there a simple way of doing this without transfering the user to a parking lot? Thanks, Jon.

Hello all, Can anyone help me out with this issue ?? I got ASTCC running, but the audios doesn't match my needs (currency, etc.). is there any way to create my own audios and replace the current one?? Thanks. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Derek, I'm not sure your recording will match with my needs. I wanna be able to do this myself with our currency here. Can you just tell me what to use and how to use it ?? Thanks. Daniel. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Derek Conniffe Sent: lundi 7 f?vrier 2005 11:59 To: Asterisk

I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com

...nux, installing Asterisk, and writing some basic conf files took about 2 weeks in my spare time. Most of that time was spent learning about asterisk, and what I needed to include in my conf files. My initial conf files were mostly adaptations of others that I found around on the net. I bought two radioshack single line phones (one was cordless), plugged them into the TDM400P. After getting the drivers loaded, and asterisk running, I ran into my first problem. I've covered this problem extensively in earlier posts (subject: "TDM400P??"), so I will just briefly mention it here. The Pro-...

...00JB%(70) VGA ASUS|V8170MAGICII/T 64M MX440SE(58) CD ROM 56X|AOPEN CD-956 RTL(22) Besides the above parts, i have 2 X100P FXO PCI cards, and 1 TDM40B I bought 2 phones from radio shack: 1: "trim phone with caller id" this is a standard phone with caller id, no wall plug - $14 2: 2.4Ghz radioshack single line cordless phone with caller id (looks similar to uniden phones) The system has a minimal server install of RH 9 with all patches There seems to be a problem with the (not "my") TDM400P hardware. Here is the origional problem: Sometimes there is a clean dialtone. Sometimes t...

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio

I haven't seen much activity on the list concerning Digium's IAXy standalone FXS device. Are they worth $100 ? Do they integrate well with * as an extension's FXS ?

Hello List, This is what I want to do: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? -------------- next part -------------- An HTML attachment was scrubbed... URL: