similar to: database queries from extensions.conf

Hi there, Regarding the calling card system, I would like to know what software tools are suitable to write the CDR system? 1) Perl (but what kind of database should be used?) 2) PHP + MySQL 3) or others suggestion? Also, which one is easier to access AGI? Any source and information would be appreciated. Best regards, Max Chow -------------- next part --------------

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Hi, Since I've upgraded to Asterisk 1.4 I can't use a MySQL database anymore for select queries :-( I'm using dbquery from MysqlPool Application 1.4 and selecting something from a table returns nothing even if I try to do a query like "SELECT 1;" Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? For information, I'm using

Hi! Is it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Hi! What are the typical ways to work around the 64 groups limit? thanks klaus

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

Hi! I just found out that Asterisk (1.4) does not write CDRs if the incoming call was not forwarded but handled internally without answering the call. E.g.: [from_pstn] exten => 997,1,Answer() exten => 997,2,Playback(tt-weasels) exten => 997,3,Hangup() exten => 999,1,Playback(tt-weasels|noanswer) exten => 999,4,Hangup() For incoming calls to 997 a CDR will be written, but not

Hi! I use the following condition: if (${FOOBAR}=YES) { ... } The problem is, that if FOOBAR is not defined at all Asterisk generates a warning: WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: =YES Of course I could use the following code, but this bloats up the code: if (${EXISTS(${FOOBAR})}) {

Hi! I've went through the READMEs and could not answer this question: During installation, the Setup program asks: Would you like update/upgrade wanpipe drivers? (y/n) For a pure Asterisk TDM installation - is it required to patch the kernel or is this only when using the sangoma cards as WAN router? regards klaus

Hi! All the AEL examples have a semicolon after the closing curly bracket, e.g: context test { 1 => Hangup(); }; but without ; it works fine too, e.g: context test { 1 => Hangup(); } So - what is the reason for the ; after the closing curly bracket? thanks klaus

Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE

Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? thanks klaus 0000:0a:03.0 Network controller: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) Subsystem: Eicon Networks Corporation Diva Server 4BRI-8M Rev

Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc & res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure / recompile asterisk after a ./configure --with-odbc=/usr/lib/ I get

Hi! I am looking for a tool (application or webinterface) which shows me the current status of an Asterisk server, e.g.: - Status of the SIP peers (registered/offline) - current incoming and outgoing calls - start-time, numbers, some history - history (calls stopped in the last 15 minutes, who hang up?) - should be possible to link those calls to the relevant SIP peers -

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Hi! I have the following weird problem: phones A,B and C are in the same callgroup/pickupgroup. A call B, B is ringing, C calls *8. Now, B is CANCELed, C gets 200 OK, but A is still in Ringing. Is there anything else I have to configure? thanks Klaus