search for: tielxu01

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 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 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 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: I have question about connection/disconnection information from Asterisk. I want to do some call monitoring selectively, that means I have a list of call cording request, I'd like to send monitoring command to Asterisk when the call connects to a specific agent who is under recording request, I need to have the connection information, similarly I need to notify Asterisk to stop

Hi All: We have cooperation PBX as NEC, I am wondering how to replace our legacy ACD by using Asterisk, but not losing tie line connection to NEC. Does anyone have any idea how to make it work? Many thanks, Tilein

Hi All: I got questioned about QoS of VoIP, I am thinking that I can configure Asterisk server in this way: I install FXO cards to talk to PSTN as carrier side, I configure SIP phones for our call center agents, that means I can get good quality from PSTN, internally connecting IP phones to Asterisk for agents, please someone confirm me if this arrangement could provide good quality of voice?

All: I tested echo test by dialing *43 under Asterisk configured by FreePbx by using x-lite softphone. I could not figure out how the call is routed to context from-internal. In sip_additional.conf, I have three extensions defined as 2826, 2800 and 2801, which all are defined context as from-internal. FreePbx doesn't define any entry for *43 as an extension in sip related config files. I

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 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 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

Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Thanks for your information. Tielin Xu