similar to: Request for help designing an unusual * application

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

I have set up an application which records incoming calls (customer problems), and then places outgoing calls (handing the problems to support people). I have written some AGI programs (in C) to implement parts of the functionality. So far I am very pleased with the overall application. But I have one very annoying, very nasty problem: The C AGI is handling a dialog with a user on an outgoing

Dear R-users! I am using R 1.7.0, under Windows XP; I also have Internet Explorer 6.0.2600.0000, Norton AntiVirus 7.60.926. Our firewall seems to want to "protect" me from downloading precompiled packages for Windows. When I try to download packages, like http://cran.at.r-project.org/bin/windows/contrib/1.7/RODBC_1.0-3.zip for example, I am not allowed to and get a message like this:

I would like the ability for a sipura sip device to instantly connect to an asterisk server as soon as the sipura sip device goes offhook and before any digits are pressed. This way asterisk can provide the dialtone and the dialplan. This also allows me to play a greeting to the phone before giving them a dialtone. Is there any way to do this, like possibly having the sipura device dial a

I'm running Wine 20010305. I am using the WINEPS driver and trying to direct it to various printers on various ports. It seems to output to port LPT1 regardless of the port I specify in the registry and win.ini. Here is a simplified scenario, in which I define only one printer, on LPT2. Can any one tell me what is happening? Follow-up question: Am I limited to 9 printers, on LPT1-LPT9?

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

Hi All, This is more of a hardware question that an Asterisk question so I hope this is still the correct place for the post. I know with the Linksys phones you can create a hotline by using the dial string of (S0<:number>). I have been trying to do this with a PolyCom phone but I have not been very successful. Does anyone know how to create a hotline phone with a PolyCom?

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address

hello.... I'm new to wine and tested and installed it, because I wanted to run the hotline server from windows under linux... upload, news works great but download from this maschine is not working! strange thing does anybody has the hotline server runing? do you have had similar problems with another network app? this are the errors.... imted from C:\windows\system\wininet.dll, setting

I will be out of the office starting 12/09/2005 and will not return until 16/09/2005. Dear Sir / Mdm, I'm currently on course and are not in office. During this period of time, I have minimal access to internet and email cccess. As such, I might not be able to reply to your queries promptly. I apologise for the inconvenience caused. In the meantime, for any technical assitance, please

How can I make it more readable? Name/username 601/601 123456789/123456789 voipbuster/abcd 601 = hotline 123456789 = Peter Pan only voipbuster/abcd is easy read/understandable! bye Ronald Wiplinger

Hello. I've seen several posts talking about line quality using Digium cards that are sharing IRQs or on machines where X is running but after trying all of those fixes I am still having a problem with line static on outoing calls. BTW, calls that are from one extension to another extension have no static, however, they have occasional clicks and pops. At any rate, I was wondering if

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Hi Is there any freeware anybody would recommend for showing database models and how they relate to each other? I''m picking up an app another developer left in a mess and I''m trying to sort it out. Diagrams would be a great help. Suggestions? -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this

I've already installed * 0.9.0. All calls from my GS phone has no ringing sound but the other end rings! I also checked this with my CVS archive. The problem exists in CVSs from (Mar 6) up to now. and I have no problem with my Xten softphone when calling with it. anybody can help? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear Sir/Madam, I'm request patch for samba 2.2.2 on Sun Server (Solaris) If you require futher information, Please let me know. Regards. Udomchai S. ======================================================= PKGINST: samba NAME: SMB based file/printer sharing CATEGORY: system ARCH: sparc VERSION: 2.2.2 BASEDIR: /usr/local VENDOR: Samba Team DESC: File

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hello- I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls per day max. I used the AMP Digital Receptionist to make a simple voice menu: "Thank you for calling xxxx". I did this for both Normal times and After Hours times. It worked fine. I then went to the AMP Maintenance window, Config Edit, got the "phpconfig for Asterisk PBX" page, and selected

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

Hi, I have a server with 2 TE110P cards. 1 card is plugged to telco line, another card is plugged with a Hicom PBX. I want to send some call to VoIP phones and all other to my PBX. I don't known how to make my dialplan : ===========Extensions.conf========== [incoming_call] exten => 090200000,1,Goto(callcenter,100,1) exten => 022956353,1,Goto(callcenter,100,1) exten =>