similar to: Asterisk Realtime Configuration

Any recommendations on good wireless SIP phones? Thanks, Pedram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081105/6248fa14/attachment.htm

I'm going to be testing the new realtime stuff further in the next few days, and just wanted some clarification on a couple of things before I start on it. I believe I can store any config file in a external config such as mgcp.conf for example, by adding it to extconfig.conf with the below syntax. mgcp.conf => mysql,asterisk,mgcpchans Doing this will require a reload of asterisk to read

Greetings, I'm having some trouble with the realtime engines. When asterisk loads, everything looks fine, there don't seem to be any problems via notices or anything. Furthermore, cdr_odbc is working, and actively logging my failed call attempts to db through ODBC using the same DSN. unixODBC and the mysql drivers are installed from source. Here are the relevant parts of the config:

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

Hi; I have entries as below in DB, mysql> select * from sip_buddies; +----+------+----------+------------+---------+------------+--------+------- -----+------------+----------+------+ | id | name | context | defaultip | host | mailbox | type | regseconds | ipaddr | username | port | +----+------+----------+------------+---------+------------+--------+-------

Hi, Realtime SIP and Extensions are working fine but facing some problems with Voicemail. Added an entry to extconfig.conf voicemail => mysql,asterisk,voicemail_users Created the corresponding table and an entry for mailbox 201. This is also reflected in the CLI as shown below. CLI> realtime load voicemail mailbox 201 Column Name Column Value

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)

I am currently in the process of testing out realtime support for sip.conf. I have followed all of the directions that are listed in the Wiki, but for some reason this does not work. When utilizing a flat file, I am able to register endpoints without any problems, and calls can proceed. One interesting side effect that I have noticed is that when I am using realtime for sip, I am unable to see

Hi I am trying to make asterisk load config from database, so far I get the sip, extension working, but voicemail seems still looking for config file, not from the database. the extconfig.conf looks like ... sipusers => mysql,asterisk,sip_buddies sippeers => mysql,asterisk,sip_buddies extensions => mysql,asterisk,extensions_table voicemail => mysql,asterisk,voicemail_users .. the

Hello list ! Can realtime dialplan be combined with 'hardcoded' dialplan in extensions.conf ?? Does a context need completely be written or in extensions.conf or in the mysql-table 'extensions_table' ? Or can I combine the two with the 'switch'-statement ?? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

Does anyone know if Grnvoip is still in business, or what's going on with them? I had an account with them, but they no longer terminate calls. Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130913/b338d9bf/attachment.htm>

Here is my file to parse and load extensions. No wise cracks about my code.... DB.php is the Pear DB module. (pear.php.net) <?php include('DB.php'); $db_host = ''; $db_name = ''; $db_login = ''; $db_pass = ''; $db_table = 'extensions_table'; define(DBINFO,"mysql://$db_login:$db_pass@$db_host/$db_name"); $db =

Hello Asterisk-Users, For some reason my CDR records for disposition and billsec are not working correctly. I always receive a 0 for billsec and the disposition is always at "NO ANSWER', even when I grab the calls. I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22. Here is information on how I do the call: -----------------------------------------------------------------

Dear all i am new to R and using a simple linear model with 4 independent variables and i am wondering if there is a command in R that will give me the partial regression coefficients thanks Pedram Rowhani Ardekani University of Louvain [[alternative HTML version deleted]]

Hi, Jan 6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such switch 'Realtime' What does this message mean ? Something wrong with the switch statement in my extensions.conf or maybe is the module net correctly installed ? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, While trying to calling Toll Free numbers using IAXTEL, the call connects, I hear about 2 seconds of voice and then the voice drops off and I get the following error message which keeps scrolling across my console screen. WARNING[-167797840]: chan_iax2.c:5967 socket_read: Received mini frame before first full voice frame Asterisk shows that the format of this call is GSM. However if I

I have configured asterisk realtime to work with two servers and a seperate MySQL DB. Each sip client registers which server it is connected to in the MySQL DB. This works great as long as the clients are 1. On the same network 2. Behind a NAT and connected to the same asterisk server as the caller. However I need this configuration to work for "NAT-ed" clients on different asterisk

Hi Guys, We have just set up Asterisk (CVS Head) for a realtime enviorment using MySQL & Asterisk Addons. I have populated the "sip_buddies" table with the same information that is came from our sip.conf, however registration seems to fail for the softphone we have set up. Does anyone have any idea as to what I should be looking for here? I'm not getting any error messages

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at