similar to: Why does Asterisk not hangup?

Displaying 20 results from an estimated 10000 matches similar to: "Why does Asterisk not hangup?"

2009 Jan 14
8
evaluate SIP response codes in dialplan
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,
2008 Dec 23
2
why does users.conf generate SIP peer and SIP user?
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
2009 Mar 06
5
work around the 64 pickupgroups limit
Hi! What are the typical ways to work around the 64 groups limit? thanks klaus
2006 May 11
3
sangoma A102 installation question
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
2009 Feb 24
7
multiple asterisks in a server
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
2010 Sep 22
5
http://www.asterisk.org/downloads naming schema
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
2009 Nov 10
2
looking for an Asterisk supervision (status viewer) tool
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 -
2009 Jun 08
3
T.38 pass-through 488 handling problem
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
2009 Jan 08
3
AEL and };
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
2009 Feb 25
3
Asterisk with Internet connectivity
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
2009 Jan 20
5
the FXS ports of Digium and damaging if connected to Tel Line
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
2010 Feb 08
2
conferencing without DAHDI
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
2010 May 17
4
identify caller hangup or callee hangup?
Hello, you know , when a call setup, either caller hangup first or callee hangup first , the hangupcause will set to 16(means Call Clearing Causes) My question is how could i identify whether the caller or callee hangup the phone first? Best Regards! -- Thanks for your supporting, have a nice day. Sucan
2009 Aug 27
6
Measuring voice quality with Asterisk
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
2008 Nov 13
5
database queries from extensions.conf
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
2009 Aug 04
4
CDR Problem - No CDRs when call is not bridged
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
2009 Jan 08
4
AEL question: testing channel variables
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})}) {
2006 Oct 18
3
identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
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
2009 Jan 07
5
recommendation for German sound files
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
2009 Mar 06
2
SIP *8 Pickup Problem
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