similar to: looking for an Asterisk supervision (status viewer) tool

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

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, 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! 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! 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! 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 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 wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus

Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal

Hi! IIRC there was an announcement some time ago that it is possible now to make conferences without the need for DAHDI anymore - but I can not remember the name of this feature anymore, and google didn't solved my problem. Thus, any references to this new system are appreciated. thanks klaus

Hi, This is a bit off-topic, but still related to telephony. Is there a barebones TAPI driver that exists that would allow me to call up a command line with, as parameter, the number to dial. For exemple, Outlook integrates with TAPI, so that TAPI driver would allow me to call my own app with the phone number as argument. ex when clicking on 555-555-5555: the TAPI driver would call

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! What is the preferred way to make database lookups from within the dialplan? I only know the MYSQL function from asterisk-addons. Are the other methods too? (e.g. for postgresql, unixodbc) thanks klaus

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 (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 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! 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 want to detect brute-force password hacking attacks - thus if there are too many failed login attempts for a SIP account I want to "lock" this account. Does somebody have any ideas how this could be implemented? thanks klaus