similar to: RoadRunner

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:

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: --

FYI, This is from Sans Internet Storm Center ... http://isc.incidents.org/analysis.html?id=178 RoadRunner Proactive Security Scan published: 2003-01-21 Recently, we did receive a number of reports indicating proxy scans from ''securityscan.sec.rr.com'' (e.g. 24.30.199.228). These scans are part of RoadRunners reactive relay/proxy scanning. All mail servers sending e-mail to

Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from

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

Hi everyone...I'm at a dead stop setting up Samba, so let me jump right into the problem: First, the setup: I have a RH7.3 box and a W98 laptop. The RH box also has a W2k Pro partition. We have them on a 4-port 10base-T hub along with our RoadRunner cable modem. Now on to the problem: As most of you are already aware, RR uses DHCP to assign IP addresses to client machines at boot time.

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

Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ ) for incomming call notification on PC (and open url with callerid), but it does not display/pop anything :-( my config is very simple... (yaacid is successfully registered as manager in asterisk) thanks PJ * dialplan: '953' => 1. NoOp(${CALLERID}) [pbx_config]

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

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

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

I have samba setup for our department and it's working fine, with mounting drives and using printers... now I want to be able to do the same thing for all faculty with roadrunner access from home. I've put the ip xxx.xxx. of the local road runner in the smb.conf file for hosts to allow, and the machine knows about the server by adding it to c:\windows\lmhosts.sam I log in as the user on

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: