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Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten =>

ENUM lookups at e164.org return a IP route for ALL toll-free numbers. I was surprised to observe that ALL toll-free numbers get a hit at e164.org. It appears that ALL toll-free prefixes have been delegated, thereby publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and even toll-free numbers that have not been allocated. :-) See below Should I care? Even though this

Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be called 'callend': $dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)"; $res = $AGI->exec("DIAL $dialstr"); $answeredtime =

Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" "!^\\+1866(.*)$!sip:1866\\1 at tollfree.sip-happens.com!" . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u"

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

On Wed, Jun 7, 2023 at 10:46 AM TTT <lists at telium.io> wrote: > I’ve reread the documentation a few times, and what isn’t clear is whether > I need an app=X parameter in the url. In other words, can I only get > events for a single named statis app? Or can I get events for the entire > Asterisk server? > > > > The command below (without app= parameter) results in

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When there is an incoming call the phone will display two buttons "answer" and "ignore". If you press "ignore" the call is dropped instead of sent to voice mail. The following is the log: -- Called 111 -- SIP/111-00001c14 is ringing -- Got SIP response 486 "Busy Here" back from

Ok that worked. Since I have not declared a statis app called “test”, does that mean any non-existent app name on the URL will subscribe to all system events? (Or is test a built-in app name) Brian From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua C. Colp Sent: Wednesday, June 7, 2023 9:56 AM To: Asterisk Users Mailing List - Non-Commercial

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

Hello, I'm writing a AGI Perl script and use the following lines to initiate a call : $res = $AGI->exec("DIAL $dialstr"); And then : $answeredtime = $AGI->get_variable("ANSWEREDTIME"); The problem is that I need to know immediately when the call has begun. I can have the call duration & establishment time at the end of the call but not in real time when it

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack

Hi all, Hopefully, I just need a second set of eyes on this one, but I just can't figure out what I'm doing wrong. I'm using an agi script to dial a number, check the dial result, and act accordingly. The problem is that I'm not getting anything back from DIALSTATUS, or HANGUPCAUSE. Here is the relevant perl code: ===============================================================

Hi, Here is my problem. I am trying to get the Status of the call if the user picked up the phone or not. It is coming as empty. Please help. Here is my extensions_additional.conf file code: [multi-dir-callback] include => multi-dir-callback-custom exten => _X.,1,Answer exten => _X.,n,Playback(beep) exten =>

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

Hi, I have set-up astcc with outgoing sip channel. Call processing works fine but after the call tables, CDR and Cards does not get updated. At the beginning it goes to the database and fetch card details and correctly provides the card balance etc. Also it indeed write the inuse field (so writing and reading from database works fine). I've inserted a break point as such in the code;

Hi all, on sourceforge.net I added the fixup release 0.5.4 of chan_capi-cm driver. The changes from 0.5.3 to 0.5.4 are: - fixed 'group' setting according to Asterisk defaults. - use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR. - full CallingPres support added. - use mutex when debug/verbose messages are printed. - set dnid on incoming call. - catch errors in