similar to: Question Asterisk Manager

Displaying 20 results from an estimated 10000 matches similar to: "Question Asterisk Manager"

2009 May 20
2
Manager ExtensionState function
Hi, I am trying to get the extension status (weather it has dialed outgoing call via SIP or IAX2), using the following piece of code however it always returns -1 on all the extensions (valid/invalid). Am i missing something ? Any help. Thanks ----------------------------------- #!/usr/bin/perl use Asterisk::Manager; use lib './lib', '../lib'; $|++; my $astman = new
2005 Jul 18
0
why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi connected with asterisk manager
hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only
2012 Mar 01
1
using AMI and Telnet to place calls
Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available" back from xxx.xxx.xxx.xxx:5060 [Mar
2003 Dec 02
0
'Stop Now', 'Restart' problems
I'm not sure where to start looking for a solution on this. I use use Asterisk::Manager to reload Asterisk with a command like: $astman->sendcommand( Action => 'Command', Command => 'Reload' ); After a while, when I try to do a manual restart or 'stop now', asterisk will not exit. Any thoughts on where to look for a resolution? Ray -- Scanned for
2013 Sep 10
0
Setting different caller-id for second leg of the Originate
Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at
2013 May 11
1
AMI Originate issue
Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...
2018 Mar 14
2
PJSIP Originate
I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002
2004 Sep 16
3
Creating conference calls from within Astman.
Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a
2014 Jun 13
1
Need to spoof the callerid using the AMI Originate
We have several customers we need to place outbound calls for (in a single system). May have to place calls for thousands of different caller ids. Customer signs a contract guaranteeing the caller id they provide is owned by them. I have everything setup for AMI Originate and can place the calls. However, I'm encountering a problem with the caller id. The system I'm dialing through
2017 Nov 22
3
Chan Local, Originate and slin
Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin
2020 Aug 06
2
Is it possible to use Stasis to control both legs of a Local channel created using ARI?
I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>
2007 Mar 30
0
forwarding loop not detected
Asterisk 1.2.16 I have an extension "102" with a Polycom 430 I am trying to protect against forwarding loops If I set the phone to forward the line to itself, extension 102 I get the following -- Got SIP response 302 "Moved Temporarily" back from 206.83.240.18 -- Now forwarding Local/102@mycontext-b2ee,2 to 'Local/102@mycontext' (thanks to
2007 May 14
1
Difference between making a call and Originate
When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com", 'Context'=>'mycontext', 'Exten'=>'899', 'Priority'=>1, 'Callerid'=>'whatever')); It creates a screech sound when the
2003 Sep 26
1
Gastman and SIP?
I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = "SIP\3846") Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The
2007 Feb 01
2
make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed.
hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]
2010 Mar 21
1
Invalid Makefiles to install asterisk with ldap
Hello , I have a problem to install asterisk with ldap. I am doing the following: make clean . / configure make menuselect LIBS =- lldap export LIBS make ====> This is where my error #make CC = "cc" CXX = "g + +" LD = "" AR = "" RANLIB = "" CFLAGS = "" make-C menuselect CONFIGURE_SILENT = "- silent" menuselect make [1]:
2011 Jun 16
1
Web based call back
Hi, I am looking for a simple solution to do this. I wish to have the user to enter their preferred method of connection i.e. for the cheapest solution to their desktop phone or mobile phone, then plan callfile based on the number that user provided and dial to the user. Any suggestions? CK -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Jul 13
1
Trouble with originating a call through Asterisk Manager Interface
I am doing a little application to originate a call through Asterisk via AMI (Perl Asterisk::Manager). It logs in successfully, does an originate command with Exten: 0020 (which is set up to answer and wait for 60 then hang up) Channel: SIP/5101234567 at test-host (which comes to my desktop machine also running Asterisk). At the target machine I see only a CANCEL to which it immediately responds
2017 Nov 22
2
Chan Local, Originate and slin
On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of
2014 Jan 31
2
callfiles.call
hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a