similar to: asterisk callerid

UPDATE - This is really really bad - check out the paypal phishing example on my blog already using Cyrillic characters http://blog.collins.net.pr/2009/12/de-latinisation-of-web.html Please forward to everyone in a position to stop ICANN, i cant believe they didn't think of this in advance. Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at

Hello list, anyone here using Asterisk together with HTB for queing incoming and outgoing packets ? I've tried to subscribe myself to the Mailinglist of the Linux Advanced Routing & Traffic Control project, but I get no confirmation. This list seems dead. It seems my test case with HTB is not giving any noticeable results. Can I ask questions on this mailinglist ? Perhaps you can

Le Thu, 24 Sep 2015 05:32:45 +0800, Bill Kenworthy <billk@iinet.net.au> a écrit : > Look into the "panic" option to ntpd - once the gap gets to big (such > as when the VM is suspended for a few hours) it goes into freewheel > and doesn't sync - its in the ntp docs. My use case is when rebooting the host (after a kernel update, for instance). The gap is about 2

Is it possible have multiple concurrent ip numbers for a single extension? How? I am using a laptop that I move around various local and remote networks so the IP numbers it uses varies. As I am on extension '205', I want be found wherever I have a connection by anyone dialling 205. Using an openvpn tunnel works well, but there are a couple of networks I dont want to tunnel (because it

Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. Thanks for answering. Le 2015-09-23 17:34, Dominique Ramaekers a écrit : > Linux has two methods to use ntp: > > ntpdate: > It will run once at boot time to sync time. (This is probably > installed on your system) > It will not run after suspend and resume... => no correction Nope. This is not installed on my system. > ntpd: > Continuously adjusts time. The

Hi, I am using libvirt with qemu/kvm with qemu-guest-agent and suspend/save/resume linux and windows vm's. Time in Windows gets handled fine with ntp but I can get neither ntp or chrony to sync after resume without manually setting the time close to correct first. Searching has gotten me to the point where I can manually set the time using qemu-guest-agent but I cant figure out how to get

On 29/09/14 16:28, Martin Kletzander wrote: > On Fri, Sep 26, 2014 at 08:16:25AM +0800, Bill Kenworthy wrote: >> Hi, >> I am using libvirt with qemu/kvm with qemu-guest-agent and >> suspend/save/resume linux and windows vm's. Time in Windows gets >> handled fine with ntp but I can get neither ntp or chrony to sync after >> resume without manually setting

Hi, has anyone successfully used the acts_as_threaded plugin with postgresql? I''m using rails 1.0 and ruby 1.8.4 on linux. Following the screencast on http://www.railtie.net/articles/2006/02/05/rails-acts_as_threaded-plugin , I got to where we''re ready to create our first post, having made the changes to controllers/posts_controller.rb, views/posts/_form.rhtml,

Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

After saving a particular VM running WinXP, any attempt to resume it (even when no other VM's are running) generates the following error: olympus ~ # virsh restore /var/lib/libvirt/qemu/save/WinXP.save error: Failed to restore domain from /var/lib/libvirt/qemu/save/WinXP.save error: internal error: Failed to reserve port 5908 This started sometimes towards the end of last year with only the

Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the

Hi I am trying to setup asterisk at home, I have 1 in bound VSP (I have a register cmd setup for that in asterisk). At home I have a cordless phone with 2 line capability - I currently have 2 spa3102's in place to handle the 2 lines ( I am in the process of buying tdm410 to handle to handle this and the backup pstn line). I also have 2 laptops setup with soft sip phones. What I would like

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of "softfax" that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp