similar to: php programming for working with asterisk

Hi everyone, Using Asterisk 1.6x here with a TDM PRI. I have to run a campaign for about 5000 numbers and then put the call to agents right away and pull up the CRM based on the number dialed. So, I am going to be doing some PHP+Ajax work. I am familiar with spool files but I don't like the fact that I can't read the status of the call in real-time. However, I know that it's the

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

Humm... Seems like my message didn't make it. Here we go again.. /Johan -------- Original Message -------- Subject: Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer <lists at jttech.se> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> On 2011-06-05 19:54, virendra

Hi all, I'm required to send a dialplan variable with every AMI event triggered for the duration of the call. For example; ... exten => s,n,Set(MyVar=${ODBC_GetSomething(${EXTEN})) ... so can I have this Variable MyVar attached in all AMI events for this call ? I can understand that untill this variable has not been set some value it may even be empty but as soon as its set I expect some

Hi list, What possibilities are there in asterisk to send an *alphanumeric DTMF*from/to asterisk !? Regards, Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120228/e62e7890/attachment.htm>

I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,

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Hi, I used Asterisk 1.8 and I have a gsm modem with 8 port. When I called target number, gsm modem and asterisk show me one of these ports active. after hangup, the actived port is going to idl status and ready to use. but after some call from extension, when I want to call another number, asterisk gives me Busy status, however all ports are idle and ready to use. I think asterisk have to

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)");

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

Hello everybody, I got tree UPS devices connected to a Debian server, and i am searching for a way to configure them. The server power-cord is only connected to one of the ups devices, and should monitor the other ups systems to sent email status. I got a HP R3000XR UPS with has a RS232 communication port that I have working perfectly. Then I have two APC SmartUPS 750V devices that are

Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why

Hello, I'm using AGI scripting with asterisk and need to execute certain commands just after Dial(). But once dial command is executed, further commands/instructions are ignored. $agi->exec("Dial","SIP/100"); $dialstatus = $agi -> get_variable("DIALSTATUS"); if($dialstatus[data]=="ANSWER") { do something.......

Hello, asterisk version 1.6.2.7 I want to get the start time of all active calls from console, could you please let me know the best way to get it. thanks, Kamlesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111214/b462516a/attachment.htm>

Hi, currently I'm trying to extend my existing backup concept but I have issues due to symlinks. rsync has problems with symlinks. First some background information: I regularly back up my / and /home from my desktop PC and my netbook to my Synology DiskStation (500GB) using rsync 3.0.6 with this script: http://pastebin.com/m43d4e353 . This works without any error messages. On the DS

I'm not sure why this started, but apparently I'm having a DNS problem. Yesterday mail started bouncing with this error: 450 Unable to find obrien-pifer.com I think the messages eventually get delivered, but not sure. I guess I'll see if this one makes the list. I checked my domain using http://www.checkdns.net/ and it gives me a couple warnings. One is that there's no MX

Is there easy way (without copying the existing rows to a temporary location and copying back) to add a new row to a specific index location in an existing data frame? Example df = data.frame( A= c('a','b','c'), B=c(1,2,3), C=(10,20,30)) newrow = c('X', 100, 200) I want to add the newrow as the second row to the data frame df Please suggest a solution that is

Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading "User Profile Configuration Options" the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes ;Sets if the only user announcement should be played when a channel enters a empty

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

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