similar to: External Custom Extension Timeout

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

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

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

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

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 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 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 have Asterisk 1.2 and a generic Wildcard single FXO card (a cheapo from eBay). I have read about many people who have used these cards without an issue and I'm just testing to work up a new system. The problem I have is that if I call the telephone number of the line attached to that card and have configured my incoming calls to, say, forward to an extension, then it will ring that

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

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

Greetings, I'm working with a new CentOS 7 installation, moving a system up from CentOS 5 due to OpenSSL version 0.9.8e not meeting PCI Compliance requirements. However, while setting up the CentOS 7 environment one of the closed source applications is requiring 0.9.8. The software vendor has advised installing package openssl098e from yum; but I'm hesitant to do so from a

Greetings, I'm running Dovecot 2.0.9 on my CentOS 6 server, for several thousand mailboxes. Recently, I've had several reports of "my mailbox is suddenly empty, where'd my mail go?" I've enabled debug logging, but I'd like to make sure I have the best level of debug to see things like "delete message" commands? I've configured in logging:

Hello all, I'm going to tackle learning C this week, and start writing my first * add-on/contribution; assuming it's actually worthy of contributing once it's done.. I think I've chosen a hefty project for my first go round here... I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want

On 10/21/2015 2:34 PM, Eero Volotinen wrote: > Remember that rhel/centos backports fixes, so just looking version > number is not reliable way to detect security issues. > > Eero Indeed, though I can say on CentOS 5 the required configuration to be PCI compliand is not valid in apache, and httpd will not start. -- ----------------------------------------------- - Nick Bright

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

I'm still learning firewalld obviously, and I am having trouble groking the documentation to understand how to do this. I know I could do an iptables direct, but that doesn't seem like the "right" way to do it. What I'm trying to do is allow a specific service, only for a specific ip. Effectively, SNMP should be allowed form a specific IP address (the systems monitor).