search for: 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";...

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: <http://lists.digi...

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 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

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 sock...

...ctories (with coporate charting capabilities). Today, its software is mainly used to edit and display charts and directories. In directory use, it displays extensions and phone numbers with convenient browsing facilities. I now have the opportunity to ask the ISV to extend its software to include click2call facilities. But given ISV background, soft will remain independant from phone infrastructure as attendant console usage is not the widest spread usage. So bottom line is if infrastructure provides click2call, that's fine and if doesn't, it doesn't really matter. So I'm looking for...

- 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,<br><br>I&#39;m trying to use latest versions of ActivaTSP and Asttapi with an Astmanproxy-enabled 1.4 Asterisk.<br...

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? Thanks, Gabriel --------------...

...SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport From: "John Doe (101)"<sip:1060 at 10.10.5.49>;tag=heMv1HvlT7DeQxPxuqcq To: <sip:204 at 10.10.5.49> Contact: "John Doe (101)"<sip:1060 at df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language="en,fr,it" Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc CSeq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5 v=0 o=- 3...

...KDePVLlTquEfJzIk7LCMdnOHHk4wn1;rport [Aug 9 22:15:50] From: "77"<sip:770000wrtc at 178.18.90.230>;tag=sRCvFQq3gUMqkl6TKTcl [Aug 9 22:15:50] To: <sip:419 at 178.18.90.230> [Aug 9 22:15:50] Contact: "77"<sips:770000wrtc at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" [Aug 9 22:15:50] Call-ID: 6aa0db27-a37b-69ee-8641-87c5bc444d32 [Aug 9 22:15:50] CSeq: 21553 INVITE [Aug 9 22:15:50] Content-Type: application/sdp [Aug 9 22:15:50] Content-Length: 1815 [Aug 9 22:15:50] Max-Forwards: 70 [Aug 9 22:15:...

...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 features). Instead of repeating to every CTI client that for instance, a call has started or stopped, I'm wondering if I should : - broadcast the same data to every CTI client (and let each CTI client build customized view from it), - while listening to specific requests from each CTI client....

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 som...

...s0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;rport [Aug 11 15:53:47] From: <sip:770000wrtc at 178.18.90.230>;tag=SGUVL1LMdvxQrUfxprZJ [Aug 11 15:53:47] To: <sip:419 at 178.18.90.230> [Aug 11 15:53:47] Contact: <sips:770000wrtc at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" [Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec [Aug 11 15:53:47] CSeq: 58874 INVITE [Aug 11 15:53:47] Content-Type: application/sdp [Aug 11 15:53:47] Content-Length: 2301 [Aug 11 15:53:47] Max-Forwards: 70 [Aug 11 15:53:...

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

...ia: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport From: "cervenka"<sip:vr1a882 at vhXXX.example.com>;tag=RDmpGm2Mubc5xQQ8NMli To: <sip:887 at ipbx> Contact: "cervenka"<sips:vr1a882 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr" Call-ID: cf2990ba-3f12-3d9e-adb6-52889c414ed3 CSeq: 41942 INVITE Content-Type: application/sdp Content-Length: 1250 Max-Forwards: 70 Authorization: Digest username="vr1a882",realm="pbx",nonce="0edd0f1f",uri...

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