similar to: how to configure my asterisk@home 1.0.9 to do call forwarding ?

Hello list We've installed asterisk@home 2.6 at the office :) I'm trying to set call hijacking for users The way this should work is this: When call comes in, a user would dial some phone code (like *8# - what we had in the old 1.0.9 setup) and pick up the call. How can I do this for the 2.6 setup ? Can it be done from the AMP web managment portal ? Our setup uses the zapata.conf file

I'm new to this and have tried to find the answer in the discussions and docs but to no avail. I even read the posting saying the password issue has already been discussed. So, at he risk of being exiled, here goes. Question 1: I've installed 0.7 and can log into the asterisk server from windows by typing http://192.168.1.11 I can log in with wwwadmin and the password I set myself

Hi list, It's been a while since I've done asterisk stuff, and I'm wondering if there any news in the field. What do you people use today for http management of debian based Asterisk setup? Preferably something with the proven ".deb" extension. Any recommendations are welcome. Thank you, Maxim. -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ?

Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to

Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to

Hello everyone I'm trying to make asterisk get a call out using the .call system. The setup is A@H 2.6 This is the content of the file is : <<< Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1 >>> I'm

--8sD16rPb6u Content-Type: text/plain; charset=us-ascii Content-Description: message body text Content-Transfer-Encoding: 7bit Griffith, I'm forwarding the mail to r-devel. Best, Fritz --8sD16rPb6u Content-Type: message/rfc822 Content-Description: forwarded message Content-Transfer-Encoding: 7bit Received: from isildur.ci.tuwien.ac.at (root@isildur.ci.tuwien.ac.at [128.131.51.43]) by

Ok, I thought the point of asterisk@home was that it automatically detected the X100P board and configured it correctly. Is this incorrect? You still need to modify /etc/zaptel files? And not just using the AMP configurator. There is no mention of this on the Asterisk@home webpage. Can anyone who has actually used ast6erisk@home confirm this one way or the other? Thanks, Dean

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw

Using rsync to copy files on the localhost, the group is being preserved even though I have not used the -g or -a options. My source files: $ ls -l ~/working/source/path -rwxr-xr-x 1 blakjak blakjak 115 Oct 8 01:37 foo.php -rwxr-xr-x 1 blakjak blakjak 6285 Oct 8 01:37 bar.php My destination: $ ls -l /active/path -rwxr-x--- 1 blakjak wwwadmin 115 Oct 8 02:06

actually it sounds like a permission issue. You said you are doing it as root, but what is asterisk running as. I've found it is very sensitive, even to ownership. Make sure the owner:group is set to what Asterisk is running as before copying. Then, I've never had problems copying vs. moving - although I could imagine it might create problems in a race condition case. p From:

Hello, Information: gcc -v: gcc version 4.3.3 (Debian 4.3.3-3) os: Debian/Testing Pulled latest release from asterisk site, compiled, installed it. I have a barebones configuration: $ ls -l asterisk extensions.conf modules.conf sip.conf users.conf voicemail.conf You can see them here: http://home.comcast.net/~jpiszcz/20090418/extensions.conf

What are these dns queries for? I'd like to disable them but I cant find any obvious reference to them in the asterisk source. I'm running Asterisk 1.4.21.2 I call voicemail and immediately hang up: I called from a sip client called line1, but I have no idea where 08c5b9e0 is coming from... 14-Jun-2009 12:37:07.926 queries: info: client 127.0.0.1#41105: query:

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

Hi, I'm using the SPA-2000 with firmware 1.06 on the Asterisk PBX, which works great for taking and placing calls, but for for some reason I can't seem to clear the stutter dialtone by either calling the extension I'm on, or the voicemail system on the Asterisk PBX. If I call my voicemail access extension directly, It tells me I have no messages waiting, yet when I hang up, then

I'm having problems getting my aastra 480i to register with the asterisk server. I can inititate calls from the phone, but >sip show peers does not show any IP address registered for this phone. I am probably missing something stupidly simple. Anyone have an example config to share or corrections for my configuration? Asterisk 1.2 aastra 480i CT has the 1.4 firmware <sip.conf>

Hi all. YAP (Yet Another Problem). I recently posted about the correct way to deal with encrypted vs nonencrypted passwords. I've gotten that to work, but I don't think its correct. So I'd still appreciate any ideas. Now my problem is I can browse the list of shares, but can't use any of them. This is NT4, Linux 2.0.31-pre9, samba-1.9.17p1. No SP3 patch applied. I

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

>>>>> On Wed, 6 May 1998 12:14:54 +0100 (British Summer Time), >>>>> Mark Tucker (MT) wrote: MT> Please could you tell me where I can find the latest MT> version of R which will run in Windows 3.11 {with MT> Microsift's 32-bit adjustment present} MT> Ideally, I am looking for a version which will do survival MT> analysis & handle

I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): 1. Sip Config Mode: Proxy Primary Proxy IP Address: *.131 Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line =