similar to: asterisk@home newbie extensions always busy

I have a TDM400P with some FXO ports, and I wanted to connect the two POTS lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk server. Hooked it up, seemed fine, called in and it answered. The problem is when the call is hung up on, the FXO port never drops. So of course then the P75 just holds the line off hook and you get a busy. So it's good for the first

I was trying to load a FreeBSD 10 VM on my CentOS 6.4 machine, and it keeps hanging and not completing the boot. I have FBSD 9.x VM's running just fine, but if I try and load 10.x it's a no go. Attaching to the console using VNC, I see: gPXE (http://etherboot.org) - 00:04.0 C980 PCI2.10 PnP BBS PMM7FC0 at 20 C980 Booting from DVD/CD... CD Loader 1.2 Building the boot loader

How do you decide what MAC address to use for that VM interface? As I just tried to change the MAC to some other value close, like I made '52:54:00:34:e1:21' into say '52:54:00:34:e1:32', and when I try and load it in, I get the following: error: XML error: Attempted double use of PCI Address '0:0:4.0' Here is one of my network entries: <interface

Tried that as well, but this has to be something that gets set at the OS level and loaded, as if you look at dmesg output, you can see all the vnet?? nodes as the OS comes online. So the question is, what is virt-install doing that creates the needed vnet interface that is part of the bridge. I really had to kill and reload the VM just to load a second interface.. --- Howard Leadmon

Maybe my google-fu is failing me, but I have spent the past couple hours looking at how to add a vnet? Device to my KVM host running CentOS 6, and for the life of me I can't get this going. >From all my research if I want to add a device I should just do 'brctl addif br1 vnet14' if I want to add a vnet14 to bridge br1. When I do this, I get: # brctl addif br0 vnet14

This is usually a very active list, but looking at my procmail log the last message I have received arrived on: >From asterisk-users-bounces@lists.digium.com Fri Jul 29 03:04:17 2005 Subject: Re: [Asterisk-Users] How can I use MySQL in the dialplan? Since that message there has been a gaping silence, any idea what is up, as I am sure seeing mail from everything else. Actually I

?? Saw the new repository notification, and figured what the heck I would try letting it upgrade me from the current v2.2.22 release that apparently is in the Ubuntu 16.04 packages, to the new repository release of v2.3.0. ?I followed the info on repo.dovecot.org, and first it started bitching about lmtp (dovecot: master: Fatal: service(lmtp) access(/usr/lib/dovecot/lmtp) failed: No such

Hello everyone, Well here is my initial posting to the list, and I will admit Asterisk is new to me. I just got everything running here a couple days ago, so still learning the ropes for sure. OK, here is my problem. Currently I have it setup talking to a couple Cisco IP phones, and some Xten softphones, this works great. I also got an account with FreeWorld Dialup using IAX2 and that

? I hear what your saying, but if you read and follow the repo page, it says run update, and then upgrade.?? Also as a test, I did remove the old 2.2 code, and installed the new 2.3 code, and again authentication fails. ?I am sure I may be missing something stupid, but the bottom line is, how can I track down why it will not auth using PAM under the newer code, when even looking at the auth

I setup a CentOS 6 server to use with KVM/QEMU, and I am getting the following error a good bit, granted it doesn't seem to be causing any trouble. I figured I would post and see if anyone has any ideas on the issue, or if I can just dismiss it. If I look at my logging/dmesg, I see: do_IRQ: 19.217 No irq handler for vector (irq -1) do_IRQ: 3.178 No irq handler for vector (irq -1) do_IRQ:

How to change status of peers "Unmonitored" to monitored? Home users showing "Unmonitored" some display timing. Name/Username Host Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server (null) (D) 255.255.255.255 0 Unmonitored voip

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

I am running Dovecot 2.1.12 under FreeBSD, and I use Outlook 2010 with imap to connect to my server. I know I used to be able to rename mailboxes, as I do this every year at year end, but when I went to rename some mailboxes the start of this year, blamo up popped the message "CANNOT Renaming not supported across conflicting directory permissions". Then only thing that has really

hallo all, i would like to know, as i would suspect, nativ bridiging should work also, if only one iax party is behind an nat router and the other has a public ip. when i connect to iax clients, which have both pubic ip's nativ bridging is working. if one of the clients is behind an nat, the iax2 channels always get routed through the asterisk server (latest stable version from cvs) ?? i

Hi, all. The asterisk setup is working fine, receiving calls via broadvoice "initially". ? When call comes in via broadvoice number, asterisk picks it up and routes correctly, as long as the call came in within ~2 min from the previous one. In other words, as long as a call comes in within ~2 min since the previous one, asterisk will answer the call. However, if the call comes in

Hello, I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know how can i monitor the extension status? when i wrote sip show peers on asterisk Extension Domain port Status 111/111 (Unspecified) D 0 Unmonitored 1300/1300 192.168.50.111 D 5060 Unmonitored 222/222

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that