similar to: Could two Asterisk servers connect through VPN

Hi List: I can not find out an example how to store "include => context name" statement into Realtime static. Please help me on this one. Thanks, Tielin

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

Dear Tim; What is folks? Where I can find it about VPN solution? Regards Bilal > Hi, > > Greetings to All, > > Im looking for some help on configuring VPN on the Asterisk PBX that I > have hosted in US. Im currently in Middle East and as everyone knows > some countries here has taboo to VOIP. Im not able to get phy phones > registered to my PBX as they are blocking SIP

Hi List: Please someone help me to point out where I can get the idea to configure Asterisk for mysql server running on different Linux. Many thanks, Tielin

Hi List: The main concern I have is that multiple accounts of the Asterisk servers could be configured in the phone, any one server of group of registered servers (2-3) down, the phone still works with existing registered servers, or the sip phones automatically register to next server referencing the account settings when current registered Asterisk server is down. Does anybody know some

Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20)

Hi List: Can someone hlep to provide one or two examples to data entry for sip.conf under the table structure? CREATE TABLE `sip_conf` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) NOT NULL default '',

Hi, We're installing an * box next week (pbxtra from fonality) and I'm trying to come up with a solution for remote users that want a phone in their home. I need VPN and QoS capability, wireless support would be a nice to have. Ethernet handoff is fine, i don't need integrated dsl or cable modem... I've been googling and cruising the list and can find bits and pieces

Hi All: I'd like to test a pure VoIP call center set up under Asterisk, Can I use existing IP routers to get VoIP traffic from service provider to Asterisk with good quality of voice? In other words, do I have to do any hardware upgrade to make VoIP work in existing enterprise environement, we have 10g Ethernet LAN? Many thanks, Tielin

Hi All, This is a general VPN question; PPTP VPNs seem to be very easy to set up with CentOS as the VPN server and the built-in windose client, but how do list members feel about the security vunerabilities reported with the MS implementation? Specifically the 6 problems reported here : http://www.schneier.com/pptp-faq.html or maybe im being paranoid? Would any of you roll this solution out

Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if

Hi, I have setup a IPSEC VPN using Openswan to connect a Draytek router to a CentOS 5.2/Shorewall 4.2.9 firewall. The VPN establishes OK but I''m getting a problem with packets from the left hand subnet getting masqueraded rather than routed down the IPSEC VPN as though they were going out onto the net. I''ve spent the last day searching Google and so far I''ve hit a

One of my user was asking, can he use VPN to access asterisk ? What does it mean ? And its possible ? How ?VPN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/325c0670/attachment.htm

Hi All Could a VPN be used to help with SIP Tunneling and QoS issues. State 1: Two IP Networks Connected via the Public Internet transmitting VoIP Traffic Say a VoIP User and VoIP Termination Provider. Each side can put QoS onto their part, but if QoS does NOT exist between them then call quality will be bad anyhow. State 2: Same as above except a VPN tunnel is setup between each side. Thus

I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that

Hi list, I have to build vpn server for 1500 clients. No encryption necessary. can anyone please recommend me vpn server. I do not have experience on vpn. I have tested openvpn on my test setup, & its working fine. I want to check if there any other vpn server available. I have not checked but can pptp vpn be usefull? My requirement is to connect 1500 clients on vpn server. Need

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source

Hello, I am having a problem with my current SIP over VPN setup. We have a server running asterisk at our office. All the phones in the office are on the same network / local to this server. We also have two employees with home offices using SIP phones over VPN to connect to the asterisk server. These phones have no problem with calls to the phones in the office, however there is no audio

I have just finished compiling and installing Asterisk on a test Debian system. All is working well. We are now attempting to get remote offices to test the system I have installed both a SIP and an IAX client at a remote office. Then I connect to our office via Microsoft ISA firewall and the Windows XP VPN client. Neither of the softphones will connect. On the IAX softphone I just get a ringtone

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it