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[k4kaleem at gmail.com]

Hi All,

our requirement is simple, we will have CALL US button on website


when they click, we want a call generated to our FS Server via WebRTC (no need for calls from FS to Users, it will be one way only from User to Server.


With call we want to send additional data like URL of page they on, login if they are logged in.
we can get data like URL and userlogin but want to sent it with SIP call as SIP Message (Probably as USER to USER Information)  so we can pull at other end.


any ideas of achieving this
Thanks,
Kaleem

[ehtasham.malik at expe...]

Hi Which Library you are using to start a call from Website ? 
Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. )
[img]https://lh6.googleusercontent.com/yE5sw1FNhk7A8YXwHvqJ_1eoUhx6Rly7EZQn8JVLClQgesUimaC5FBhoqcjL_0TiGnF8-z4kSqJywWlkYPcySl1rLcS18hzt1PbCqtGbvbtl6TVHQlFadPziihRZ17vCkCMArghS[/img] WWW:[img]https://lh4.googleusercontent.com/4Imm7d5fvTo1pttR1BIKC8HO8j5SLyBZTB8tWImJUvl8pOlWGLXnQW8yrpknZc1LXvs6Fh7Dqnb7364OiYyUY9k5ZrAhlHaERf2ydHCwilnMDEstZGguJryPOTargVvyKGpl6nE4[/img].@expertflow.com (andreas.stuber@expertflow.com); Skype:[/url]shani.awan3











On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com (k4kaleem@gmail.com)> wrote:

[k4kaleem at gmail.com]

 Salaam Ehtasham,

we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to.


Regards,
Kaleem




---------- Forwarded message ----------
From: Ehtasham Ul-Haq <ehtasham.malik@expertflow.com (ehtasham.malik@expertflow.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: Ahmed Hasan <ahmad.hasan@expertflow.com (ahmad.hasan@expertflow.com)>
Bcc: 
Date: Thu, 25 Nov 2021 15:28:31 +0500
Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI
Hi Which Library you are using to start a call from Website ? 
Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. )
[img]https://lh6.googleusercontent.com/yE5sw1FNhk7A8YXwHvqJ_1eoUhx6Rly7EZQn8JVLClQgesUimaC5FBhoqcjL_0TiGnF8-z4kSqJywWlkYPcySl1rLcS18hzt1PbCqtGbvbtl6TVHQlFadPziihRZ17vCkCMArghS[/img] WWW:[img]https://lh4.googleusercontent.com/4Imm7d5fvTo1pttR1BIKC8HO8j5SLyBZTB8tWImJUvl8pOlWGLXnQW8yrpknZc1LXvs6Fh7Dqnb7364OiYyUY9k5ZrAhlHaERf2ydHCwilnMDEstZGguJryPOTargVvyKGpl6nE4[/img].@expertflow.com (andreas.stuber@expertflow.com); Skype:[/url]shani.awan3











On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com (k4kaleem@gmail.com)> wrote:

[k4kaleem at gmail.com]

Hi Kaiduan,

thanks for looking into it.


Verto looks cool. we have no restriction as to what to use. main item is to attach data to call so sip client at end user can strip and show data to agent.


no need for user to login to enter credentials, we want simple "call us" type button which generates a call.
to make it safe from attacks as server will be on cloud, we would like some sort of safety measure, either a login and pwd, which gets passed to freeswitch to verify its genuine call from a webpage or app. or some hidden message within the generate call command so freeswitch can verify and drop any calls which arent from right source so answering party doesnt get too many junk calls from random bots who discover port is open on Cloud FS.


Regards,
K
---------- Forwarded message ----------
From: kaiduan xie <kaiduanx@yahoo.ca (kaiduanx@yahoo.ca)>
To: "freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)" <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: 
Bcc: 
Date: Fri, 26 Nov 2021 02:57:01 +0000 (UTC)
Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI

You can use JSON based VERTO protocol instead of SIP to make things easier. Does the user have to login in FS?


/Kaiduan



On Thursday, November 25, 2021, 03:30:52 p.m. EST, kaleem rehman <k4kaleem@gmail.com (k4kaleem@gmail.com)> wrote:




 Salaam Ehtasham,

we are looking to use JSSIP or SIPJS, we are flexible and can look into SIPML if for any reason we have to.


Regards,
Kaleem




---------- Forwarded message ----------
From: Ehtasham Ul-Haq <ehtasham.malik@expertflow.com (ehtasham.malik@expertflow.com)>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Cc: Ahmed Hasan <ahmad.hasan@expertflow.com (ahmad.hasan@expertflow.com)>
Bcc: 
Date: Thu, 25 Nov 2021 15:28:31 +0500
Subject: Re: [Freeswitch-users] WebRTC calls one way with custom sip messages UUI
Hi Which Library you are using to start a call from Website ? 
Malik Ehtasham, CTI Product Manager / Technical Lead (Mr. )
[img]https://lh6.googleusercontent.com/yE5sw1FNhk7A8YXwHvqJ_1eoUhx6Rly7EZQn8JVLClQgesUimaC5FBhoqcjL_0TiGnF8-z4kSqJywWlkYPcySl1rLcS18hzt1PbCqtGbvbtl6TVHQlFadPziihRZ17vCkCMArghS[/img] WWW:[img]https://lh4.googleusercontent.com/4Imm7d5fvTo1pttR1BIKC8HO8j5SLyBZTB8tWImJUvl8pOlWGLXnQW8yrpknZc1LXvs6Fh7Dqnb7364OiYyUY9k5ZrAhlHaERf2ydHCwilnMDEstZGguJryPOTargVvyKGpl6nE4[/img].@expertflow.com (andreas.stuber@expertflow.com); Skype:[/url]shani.awan3











On Thu, Nov 25, 2021 at 4:16 AM kaleem rehman <k4kaleem@gmail.com (k4kaleem@gmail.com)> wrote:

[k4kaleem at gmail.com]

Hi All,


any takes on this plz.

On Fri, Nov 26, 2021 at 5:35 PM kaleem rehman <k4kaleem@gmail.com (k4kaleem@gmail.com)> wrote:

[k4kaleem at gmail.com]

hi Thanks for ur msg, it was chopped towards the end says: ""the issue is" and nothing after.

this gives me idea on solution.


on verto demo, i can see it asks for login and then extension to make call. if i want all this hidden from user and in background. so end users sees a button only in app and it says "call us" once clicked everything is in background like logging softphone and starting call. all users hear is freeswitch answering and playing relative IVR script.


what would be best way to achieve this.
also, from Verto side, if i want to attach some data with call. what method i need to use to achieve.


thanks,
k


On Fri, Dec 3, 2021 at 7:17 PM Support from NetworkedAudio LLC <support@naud.io (support@naud.io)> wrote: