similar to: Controlling calls

Hi, I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm using an OpenVox D410E and the drivers are loaded. My system.conf looks like this: # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED span=1,2,0,esf,b8zs bchan=1-24 # Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED span=2,1,0,esf,b8zs bchan=25-47 dchan=48 These

Dear list; I'm trying for forward some calls to an others asterisk using IAX2 protocol. But My asterisk can forward some calls and for others it hangs up automaticaly. Before my asterisk was working perfectly, i do not know what is happening!! When i try directly zoiper with my provider's asterisk it works perfectly. Here is the output from the cli when i made a call that asterisk hangs

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20)) thanks ________________________________ De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re :

Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]

Greetings folks; I have a bit of a conundrum, and I can't tell if Asterisk is doing something daft, or whether I'm clean missing out why it's doing what it's doing. So, I have a dialplan that looks a little like this: -------------------- [start] include => dids include => everythingelse [dids] ; Test exten => 8378,1,SetCallerID(3015551212) exten => 8378,2,Hangup

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten => *100*,1,AGI(test_app.pl) ... exten => 190,1,Answer() ... exten => *100*,1,AGI(never_reached.pl) ... A "normal dialplan reload command" would echo no warning or

Hi, Here is a scenario: 1) A call comes in on an outside line on a DAHDI device 2) The call is answered by a SIP extension (Linksys SPA942 to be exact) 3) The SIP extension places the outside call on hold 4) The same SIP extension dials another extension. Is it possible for the dialplan in step 4 to discover the channel of the call placed on hold in step 3? In other words, since it is the

Hi all, Is there a way to get the "called id" (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $callerid = $input{'callerid'}; $AGI->say_digits($callerid); } Thanks in advance, Angel.

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir

Hello, We want the ability to choose from an AGI script whether or not to enable faxdetect for calls over SIP or DAHDI. Is this possible, or can anyone suggest a workaround? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was

Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels. I want to use the teleyapper system for broadcasting call for security reason but i need that all channels are free when a security call is ready to start! I already search in the old post without success. Can anyone help me? Thanks and sorry for my newbie english -------------- next part -------------- An HTML

Hi. I've got a working dial plan on my home system but there are problems with it and I was hoping someone more comfortable with dial plans might be able to help. In a nutshell here's what I'm currently doing on an incoming outside phone call [default] Set(TIMEOUT(digit)=3 Set(TIMEOUT(response)=60 exten => s,1,NoOp(Answering in default context) exten =>

On Mon, 18 Jan 2016 16:09:17 -0200 "Ethy H. Brito" <ethy.brito at inexo.com.br> wrote: > On Mon, 18 Jan 2016 09:38:52 -0800 (PST) > Steve Edwards <asterisk.org at sedwards.com> wrote: > > > On Mon, 18 Jan 2016, Ethy H. Brito wrote: > > > > >> how to flush user input before READ()? > > > > How about a read() to a dummy variable

Dear R experts This probably seems very easy to you guys, but I'm a beginner and would be really glad if someone helped me with this: I am trying to automate the execution of an R script (let's call it "myscript.R") by passing a variable from a bash script to myscript.R. I know I can use the command Rscript, but I don't know how to declare in bash which variable will be

Hi there, I have a RoR app but for one of the funcationality (photo uploads) I am using PHP. I need to know how I can share/access session information from RoR in PHP since the PHP also updates the database and the entry information is contained in the RoR sessions. Please advice. Thanks in advance. Adil --~--~---------~--~----~------------~-------~--~----~ You received this message because

I've got an agi that recognises speech (via Google) and another that turns text into speech (tts) (via Microsoft Translate). Both are web APIs, both called via seperate python AGIs. I've googled and I'm probably missing something pretty newbie 101 here, but is there any way, or fiddle, that I can play some audio to let the caller know that their weather forecast is being fetched,

Hi Gang I have stumbled of this problem. I need the P-Asserted-Identity header in an AGI scrip. In the Dial-Plan I do: same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) In the AGI I do: my $pai = $AGI->get_variable(PAI); This works fine, unless the PAI contains quotes: P-Asserted-Identity: <sip:1000 at 1.2.3.4:5060;user=phone> I get "<sip:1000 at

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");