similar to: Asterisk with Internet connectivity

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! 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! 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 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! 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! 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 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! I have some (short) question about the Eicon DIVA (V-)4BRI cards I want to have (short) answered before buying the DIVA card. I know there are several Eicon guys active on the list, thus I ask on the list instead of directly to Eicon so that all other will benefit as well. 1. Do Eicon DIVA (V-)4BRI cards support TE and NT mode? 2. Can the clock (master/slave) be configured independent

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! 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