similar to: Stumped on vMail problem, any ideas?

Hi All, I have registered "abhijit" for SIP in asterisk Server. I am able to register my softphone (SJPhone) to the server using the name "abhijit". But whenever I try to make any calls I am gettinh the following error message:- *CLI> -- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper:

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

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

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

hi list, I'm running a newly installed asterisk@home an i registered two soft phone. both soft phone are registered 8901/8901 x.x.x.x D 255.255.255.255 50710 Unmonitored 8900/8900 y.y.y.y D 255.255.255.255 6281 Unmonitored but when I call one another, they are always busy and directed to its voicemail Sorry, if this was posted before TIA

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

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

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

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

Hi, I'm still in the process of debugging this, but I have a gotoif statement that looks like this: exten => 26,1,GotoIfTime(7:00-18:00|mon-fri|*|*?ext-queues,210,1) exten => 26,n,Goto(ext-local,${VM_PREFIX}127,1) I have others setup the same way that also seem to have the same 'issue'. The issue is that they work, but they seem to require (and I don't understand why) a

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 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

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

All temporary files are kept in a single directory. At the end of the autotests that one directory is deleted. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- autotest.sh | 20 +++++++++++--------- 1 files changed, 11 insertions(+), 9 deletions(-) diff --git a/autotest.sh b/autotest.sh index c9f8a2d..d658cf3 100755 --- a/autotest.sh +++ b/autotest.sh @@ -40,6 +40,7 @@ # an

Hi, I ma using Asterisk 1.2.18 & FreePBX 2.2.1. I have assigned every users in office with Polycom with 2 extensions as below 555 8555 I have configured Follow-me to ring when the users doesn't picks the phone on line 1(555) after 10 seconds & then ring the line 2(8555). But this is not a standard telephony which I have been advised to change like below. If someone calls