similar to: howto dump binary data on zap channel?

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! 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! 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! 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! 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! 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! 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 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

2009/3/4 Klaus Darilion <klaus.mailinglists at pernau.at> > core show function SIPPEER Thanks : that's exactly what I was looking for !! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/7d05f172/attachment.htm

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'm trying to build a SIP / ISDN BRI gateway. I'm using asterisk and zaptel 1.0.6 with the bristuff patches. I have a Billion HFC card connected to a BRI ISDN line. Unfortunately each time asterisk tries to make a call on that channel, I get NOTICE[6236]: app_dial.c:759 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time When I try

I new in asterisk world so, please, forgive me if I say something stupid. At least, and after a lot of tryes, the isdn card seems to be registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo

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

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

Hi, I have a working Asterisk 1.2 server on kernel 2.6.22 with the OSLEC echo canceller on a Digium PRI card. I recently switched to kernel 2.6.23 with the MG2 echo canceller (nothing else changed). Each time I try to establish a call on the PRI line I get a congestion signal. in /var/log/asterisk/full: Feb 20 08:09:43 VERBOSE[10657] logger.c: -- Executing

Watch the output of 'pri debug span 1' on the Asterisk server while placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) might be relevant. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andy Goss > Sent: Monday, October 10, 2005 5:58 AM > To: Asterisk Users

Hi, I have an asterisk connected with PRI (Zap channels). If I try to call a number, and recieve cause code 27 because the line 553192 is out of service, but the call continue...is it ok? Here the console messages -- Executing [98 at TRONCAL-PRI-76:5] Dial("Zap/1-1", "Zap/g1/553192") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/553192