similar to: YAACID isn't working

Hi everyone, i'm announcing YAACID: Yet Another Asterisk Caller ID V0.9 it's the first public release. we've been using it in our office for a couple of months now. YAACID is a native Windows (.NET) program that sits in the notification area and logs into the manager interface. it waits for a call to come in on a monitored channel and then pops up the callerid info in a very

I use SIP over VPN with RR from TWC no problem, connect via WiFi. According to http://www.speakeasy.net/speedtest/ I am getting 3.5Mbps down and 353Kbps up at this time (6:15pm Saturday). My laptop currently has an X-Lite (free version) softphone with GN Netcom USB professional contact center headsets (GN8110 USB XP adapter). We have found that the headset makes a major difference in the quality

To All If anyone can shed any light on this it would be greatly appreciated. My phones are unable to enter pins numbers correctly when required by the party they are calling. For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if

Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

hey all, from a lot of great feedback we found some bugs in YAACID that appear in *@home and the stable asterisk versions and older CVS versions. we're putting the fixes in and will have a new one tomorrow on the web site.. Thanks, Mark

Hi - I am going to open port 5038 on my firewall so that I can use YAACID to spawn browser popups on an incoming call. My question is, under manager.conf, what are the suggested settings so that I can get the browser popups only? I'll be at different IPs so I can't lock it down that way.. I guess I don't need any write access? [managername] secret=secretword

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

Hii, I want to forward calls from an asterisk server to a local gsm network. I have read the wiki pages on various forums. But the thing i want is to receive the call(Voip) from an asterisk server then it should be forwarded to a gsm network & again to either a gsm/ PSTN from the gsm network itself. Please post a help. Thanx in advance. -- Day by Day in Every Way I'm Getting Better

Does anyone know of any companies where I can interconnect with for SMS? ? .o-------------------------------------------------------o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

Hi, I currently use? Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on a debian sarge with a kernel 2.4.27 on a P4 3Gig with 1Gig of memory When i use i4l on any call, the called party ( on the telco operator side ) ear me with a delay of 1 sec after 1 minutes , 2 sec after 3 minutes and so on... After a quart hour, the delay make the conversation just

Hello all, Recently I migrated all telephony in my house to asterisk thanks to the Asterisk, QuadBRI which works wonderfully well. Some small tweaks to make but that's on the long list. On the short list is the ability to reliable send and receive SMS. For SMS I already built a script email2sms, but sometimes the SMS doesn't get send from some reason, the sms log then reports something

For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore

HooDaHek 0.6 has been released. So soon, you say? Well, the best laid plans of mice and men... Steven BerkHolz is a pretty sharp stick and said to me, "Why don't you have HooDaHek change the CallerID when it looks up the name in the database on an incoming call?" Much head smacketh ensued, and as I made that change for Steven, I noticed that I had the way wrong version of

Hello, i wonder why i didn't find a solution for this problem yet, because it seems very common: I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some SIP-Softphones which i can call from outside by calling the phonenumber of the Asterisk-Server and then dialing the number of the SIP-Phone. If I make a call from a SIP-Phone into PSTN, only the MSN of the asterisk-server is

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.

Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I've signed up for a NuFone account, and added the following instructions to my config files per NufFones directinos: iax.conf [NuFone] type=peer host=switch-1.nufone.net secret=password extensions.conf (under the [default] context) exten => _1NXXNXXXXXX,1,Dial,IAX2/f00b3r@NuFone/${EXTEN} I then get this message in the CLI: -- Executing Dial("SIP/jake-fe5d",