similar to: Different click2call?

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,

Dear List wanna configure click2call in such a manner that my asterisk box call two mobile numbers and connect both numbers to talk. I have configured voip gateway, my internal and external calls are working fine. please help , abhi -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

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

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

- ActivaTSP can't work with Astmanproxy as Asmanproxy needs to be patched, - Asttapi wouldn't terminate a completed call. Which option would you pick ? Is there any other option (free or commercial) for Outlook click2call ? Best regards ------=_Part_283_12644120.1196417210166 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline

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?

Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call

Hi, and another interesting bug report, where I'm not sure what the correct behaviour of openssh should be. Thanks for your comments: > I find myself frequently using OpenSSH to log in and perform a single > command (particularly in a script to perform some quick, simple task on > multiple machines I administer). If the '-t' option is not included, the > behavior is no

Hello I'm trying for several days now to get ICE support for my Asterisk 11.23 on CentOS 6. My call setup : sipml5_webRTC (nat) --> public Asterisk on 178.18.90.230 --> softphone Zoiper (problem : no audio) Reverse does not work either. (problem : failed get local SDP) I followed this guide : https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

Hi, I'm evaluating possible design for a specific CTI application. CTI client is a fat client (it's a customer's requirement) which exchange data with a CRM server (build on mainframe). CTI client must : - display custom view mixing ongoing calls, presence and some user preferences (such as this user has forwarded his calls to his voicemail) - request call origination (click2call

Folks, I am curious to know what the best way to build click2call with asterisk? There are a bunch of examples of the web that use socket to launch first leg of the call and then dump the call to a context that dials the second leg of the call. Unfortunately, none of the solutions I found explained how to get the call status of the first leg. What if there is some issue with the channel, what if

Hello Using Asterisk 12.8.2. I now have the "via ICE" messages in the RTP debug (see below). If you look in the SIP debug (see below), you also now see the "ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the webRTC client. But still no audio ! None at all ! In both directions. You can see in the SIP debug that the IP-address in de

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

Dear r-helpers, Here I would like to have your kind helps on generating Venn diagram. There are some packages within R on this task, like venneuler, VennDiagram, vennerable. But, vennerable can not be installed on my Mac book. It seems VennDiagram can not work on my data. And, venneuler may have generated a wrong Venn diagram to me. Do you have any experience/expertise on those Venn diagram?

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Jerry, Everybody: 3.0.25c compiled from source on a mandriva 2005le server in my no (AD, LDAP, Kerbose) environment. A full day of production and nothing more that a whimper out of the system. So, at least in my case 3.0.25c looks good.