similar to: Asterisk 1.2 1 FXO Problem

For some reason my voice mail stopped working properly. I was able to go in as a new user, set the password and options and now can never log back in using the password I assigned the mailbox. I can log in through the web interface with that password fine, and the voicemail.conf looks fine but every time I try to check messages I get "Password incorrect please try again" until it

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following

I have a Polycom IP501 phone and have set it up to download the config from an FTP server, it did this once and now is in an endless loop of trying to contact the FTP server, failing, then rebooting. When I watch the FTP server logs it looks like the phone starts a session, ends it, starts it, ends it until the phone reboots. It is annoying like nothing I can describe! I have tried Windows 2003

Is there a protocol I'm supposed to use here? It seems that people are asking 100 questions a day and SOMEONE is helping them, and I've posted this three times and not even an "I Don't Know". My third repost: Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has

Hello, I'm having trouble getting this to work: I have a ring group that dials an extension and if no answer dials a cell phone. If the cell phone doesn't answer I want to go to voicemail or another extension. I have set the timeout to 15 seconds but it never actually works, it will just ring until the cell voice mail picks up. I'm using Asterisk@Home 2.8 and a TDM400P card.

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

hello I am running Samba 2.2.8 (latest) that I downloaded the binary from the Samba site. I have this problem in older versions of Samba as well. I have under Redhat 8.0 security level = user works fine under NT/2000/XP platforms, but when I try to login from a Windows 95/98/98se/Me machine, samba prompts me for a password to resource \\netbiosname\IPC$ how do I prevent this from happening in

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

I'm not exactly sure how to phrase this, but I have two offices that I want to each have an * server. One office has two POTS lines coming in, the other has five. I'll call the two line office A and the other office B I want office A to be able to extension dial office B, which I think is straightforward enough because I'll just add a line in extensions.conf that points a particular

I have set up two Asterisk peers to use each others dialplans using 7+ the normal dialplan of the other server (i.e., 7101 dials extension 101 on the other server). I want to be able to ring an extension locally AND remotely at the same time when a call comes in but I don't know how to set up * to do a dial group that rings a remote server. So, if someone calls in it will ring extension

I have been having a strange issue with my Asterisk 1.2.1 server. I have a TDM400 for the three POTS lines I have and I can receive calls without any problems. But sometimes (not everytime, but 70%) when I dial out of those lines it drops a number and of course gives me the telco error message. How do I know that it's dropping? Because I thought it was a line problem and the tech came out

Hello All, I have the latest Asterisk and am using the TDM400 card with 4 FXO ports. One problem I have is that I want one of my Queues to go to a cell phone if all agents for the queue don't answer. The problem is that once the TDM400 goes to a POTS line it does not time out after 15 seconds like I have configured (and like the other agents), it will ring to voicemail which doesn't

ok, I fixed the file permissions problem using chown... which is good. now, I am still having the problem in WIndows 98/Me. samba keeps prompting for a password to: //netbiosname//:IPC$ what in the heck is :IPC$???? it works fine in windows XP or 2000, so what am I doing wrong that it doesn't work in 98/Me? I tried reinstalling windows me on my daughters computer and running the windows

Has anyone had any echo issues with the Digium TDM02B FXO card? I purchased a clone wildcard on E-Bay for $8.00 and have had horrible echo issues.... I'm assuming it is because of impedance issues? Just wondered what people's takes were on the TDM02B FXO and echo?

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

Is there any point in reporting specific devices that work with umass(4) any more, or is the assumption these days that anything that will work on a Windows or Linux box, without additional drivers, will also work on FreeBSD? The cheapo 'pen drive' that I picked up on ebay yesterday surprised me by just working, with no pain whatsoever... I'm wondering if maybe I shouldn't be so

I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different business. I know that for a DID the routing is simple but I'm not seeing where I can match up a DID with a Zap channel. I'm currently looking into the zapata.conf file

Anyone familiar with ePrompter? It's a nice little e-mail checker that floats in the system tray. Anyway, I have it running under Wine, it appears to load fine and everything, but for some reason it can't seem to connect to the internet. I don't have a lot of experience with software config under Linux, and I know almost nothing about Wine...does anyone have any suggestions on this?

Hi all, I have done a major upgrade, first I have bought a Digium TDM400P (TDM02B) 2-Port FXO to replace x100p cards, than moved from an old version Asterisk to a new centos 4 box, compiled asterisk from Asterisk CVS-HEAD built on 2005-06-11. Seems that my iaxy is now "incompatible" with the TDM400P. When I receive a call from the TDM02B, the caller can't hear my voice (I am using