similar to: click2call for conferencing two mobile numbers

Hi, I have a PHP code with AMI to using in click2call system. here is my code: $user = "usernamr"; $secret = "secret"; $channel = 'SIP/' . $sip; $context = "from-internal"; $waitTime = "20"; $timeout = 20000; $priority = "1"; $maxRetry = "2"; $pos = strpos($number,

Hi there, Just to announce that I've improved upon a greasemonkey script which allows users to dial any number (in the given regex format) by turning it into a clickable hyperlink. The script uses greasemonkey's ajax callback to a simple php controller script, so that the click does not navigate away from the current page. It requires an Asterisk Manager connection. See

On Monday 20 Dec 2010, Olivier wrote: > 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk> > > > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application > > (written by someone else before me) which sets up calls by creating > > files of > > the general form > > > > Channel: SIP/$INSIDE_NUMBER > > Context: $CONTEXT >

Hi guys, I've being trying to create a 'click2call' for internal use in the place I work. The idea is pretty simple and actually I've a simple click2call working working already... Well, my question is: do you guys have any tip in different ways to create a call in Asterisk using AGI + PHP? Right now I'm only using simple PHP and sockets to talk to the Asterisks using the

Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional

Hello, i have asterisk 1.4 installed and i want to use click to call in order to do an outbound call if there is any php code in order to do this operation thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110519/417ac394/attachment.htm>

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

*Dear List* Plz help, i am not much experienced with asterisk. i configured it on ubuntu 12.04. no problem when i call any mobile no(0091XXXXXXXXXX) but when i call on my local asterisk no.(101,102 or 105) it is not connecting giving error "Auto fallthrough, channel 'SIP/lucknow-0000006f' status is 'CHANUNAVAIL' *while when i call 200 it is playing audiofile successfully.

Hello Experts. Im working with Asterisk PBXand freeswitch PBX. I have a challenge with FXO card in Asterisk and i could not solve it yet. I hope you could guide me in this regards. When i want route the call to FXO channels, Before the callee answer the phone (pick up phone), The channel is answered with FXO card. How can change this treat so that the callee dont answer the phone, the channel dont

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

Hi all, I'm trying to send an email with the voicemail details and I want to send a HTML link on it to make a click2call to the voicemail main, but the email is send with 'text/plain' encoding and thus it will not show the link, but the HTML in plain text on the body of the email, How can I change the enconding to 'text/html' so the link will get displayed correctly?

Hello List, We are deveoping apication/system based on PHP5, Postgre,Ajax,ect.. It should be compleate sistem for realystate agensy and road worers(agents) and it will be distributed system. We made very good inplementation based on asterisk and OSP for distributed offices and it will be part of system (integrated). We would like to implement some options in system: Agent have contact list with

Hello, iam searching for an Firefox plugin which can make an sip Invite and Redirect after 200 OK, so i dont have to use a softphone, just to initialise a call by clicking on a number i've found some plugins which only works with a softphone installed on the system but nothing which works good with asterisk. my other problem is that we use firefox 3.5 mostly on mac so maybe there are

Olivier, I think you are getting confused. Call me on 212-203-4357 and I'll answer your questions but basically I think you are doing this the wrong way. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ________________________________ From: asterisk-users-bounces at

Hi, I am not been able to contact with mentor of this project. Can someone else from the community help me with this ? Regards, Anusha Srivastava On 3 March 2018 at 11:16, Anusha Srivastava <sidd.sinha89 at gmail.com> wrote: > Hi Martin, > > Any update on this ? > Regards, > Anusha Srivastava > > > On 28 February 2018 at 23:37, Anusha Srivastava <sidd.sinha89

I'm not sure if this is an Asterisk thing, a handset thing or a telco thing, so please be gentle with me if this is not the right place to ask ..... When placing a call over a SIP channel to a mobile phone, if the phone is engaged, it does not return a BUSY status straightaway. Rather, I get a ringing-out tone for the timeout duration specified in the Dial() statement; *then* I get

Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I would like to participate in GSOC 2018 with Xorg to contribute to project "Initial Nouveau Vulkan driver' I would need some help in how to get started with the same. Regards, Anusha Srivastava

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on