search for: set_variables

Hi all, My VoIP provider (race.com) doesn't send name info with CallerID, so I wrote an AGI script that does the following: 1) If it's a toll free number (800|888|877|866), set the CallerID name to "TollFree Caller" 2) Use curl to look up the number in Google phonebook 3) If a business listing, set the CallerID name to business name, as is. 4) If it's a residential

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

...set a somewhat sane default size diff --git a/vivified/ui/main.c b/vivified/ui/main.c index 617e317..fe4b35c 100644 --- a/vivified/ui/main.c +++ b/vivified/ui/main.c @@ -67,6 +67,7 @@ setup (const char *filename, const char vivi_application_set_filename (app, filename); vivi_application_set_variables (app, variables); window = gtk_window_new (GTK_WINDOW_TOPLEVEL); + gtk_window_set_default_size (GTK_WINDOW (window), 400, 450); g_signal_connect_swapped (app, "notify::quit", G_CALLBACK (gtk_widget_destroy), window); g_signal_connect (app, "notify::filename", G_CALLBAC...

hi everyone, actually i want to release an IVR system using PHPAGI API , in this IVR i want to get value from the user. I already used get_data defined in phpagi but they are not able to get the value given by the user and store it in a php variable. i tested this : $result = $agi->get_data('beep', 3000, 20); $keys = $result['result']; but every time i found in $keys variable

Hi, I am pretty new to Asterisk. I am trying to make sure some human being answers the phone not the voice mail machine. How can I programmatically identify that? Here is my Sub: sub DialPhysician { my ($self, $con, $PhysicianPhone, $call_id, $conv_id) = (@_); to_log($self, "Inside Dial Physician", 2); my $DocPhone = "1".

Hi, I am attempting to connect to the blacklist database using PHPAgi and it always seems to hang. The code snippet I am trying is: $r = $agi->get_variable("CALLERID(num)"); $cidnum = $r["data"]; if ($cidnum < 1000000000) # No valid callerid. { exit(0); } $r = $agi->database_get("blacklist", "$cidnum"); if ($r["result"] ==

OK, I need to dial a macro from AGI and needs to pass an argument. Ok, I found an bug report, but it was stated "un fixable?" really after 5 years? https://issues.asterisk.org/view.php?id=2470 I found this email in the archive, but no solution other then the dodgy work around? http://www.mail-archive.com/asterisk-users at lists.digium.com/msg85048.html I have

I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which plays few wav files depending on external parameters. Can I have a dial plan inside my AGI? If not, how do I accomodate user who needs to reach extension 2 from my agi? I have tried stream file and get data but the two commands did not work at all.

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Here's something that should be easy for RUBY pro's. Here is a script: 1.times do r = $agi.exec('DIAL', SIP/voipuser&Zap/32&Zap/33&Zap/34&Zap/35) r = $agi.get_variable('DIALSTATUS') # $agi.set_variable(' WHOANSWERED

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

Hi All, i have created one prepaid PERL AGI script to integrate asterisk users in our current Oracle Billing System. I am using $AGI->exec('Dial', $dialstr); in script after getting the MAX time out for the priticular call. But when the channels increase on my asterisk more than 50-60 asterisk get crashed and i am suspecting the cause is of AGI Script. because when i check ps on

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

I have a need to have a GUID (for example, bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the "-"]) generated in the dialplan. Is there any asterisk function that would do this ? I would prefer not to have to shell out every time a call comes in. Julian

Hello list, I just came into a strange problem wth astcc. the trouble is astcc.agi does not bill some calls. The calls are logged in the cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an empty dstchannel, and with a lastapp field of "hangup". I suppose that astcc.agi was not able to get the answeredime variable from the SIP channel... I have added a few functions to

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in hangup handlers ? I'm planning to store some RTCP stats. Saving them in CDR(userfield) would be

I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx-xxxx or xxx-xxx-xxxx. Is there any way to parse characters out of the dialed phone number so that I only end up with digits (remove spaces, parenthesis and dashes)?

Hi, I've been fussing with this for a while now - and cannot seem to get it to work correctly - or rather as I desire it.. I'm trying to implement a 'find-me' feature in my voicemail. Basically - pressing 1 at the voicemail puts the caller into a queue. The members of the queue are my cell phone and my work DID.. I need the queue to call both numbers at the same time and wait