Web embedded sip phone jobs

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10,382 web embedded sip phone jobs found, pricing in USD
BULK IVR (BULK SIP CALL) 6 days left
VERIFIED

We need an API to make bulk calls over SIP server. API details listed below, *Minimum 500 concurent calls *Text to speech (Turkish & English) and audio file (m4a, wav, mp3) call *Simple IVR functions >Prepare scenario (first message, 2nd message etc) >Get pressed digit after a proper scenario node >Basic equal to opeartion according

$982 (Avg Bid)
$982 Avg Bid
13 bids

Hello, I need to have an opensips server installed with cgrates. This will be for SIP trunking with nat support. I will need to be able to route inbound DIDs to customers. Also will need to be able to setup accounts in cgrates so that we can have about prepaid calling on the account. I think we have several other project that you might be able

$536 (Avg Bid)
NDA
$536 Avg Bid
3 bids

We are an Shophosting provider for Sellers who wants to create their own Online-Store, without any programming ...we start with some small projects now to check who is the right future programmer for us. project description: Update our internal used Editor with the fuction of embedded Youtube Videos Have a look to the attachment for a details

$28 (Avg Bid)
$28 Avg Bid
18 bids

...Script to do the following: - redirect embedded links in PDF file - store event details passed from PDF link into MySQL DB - include tracking of PDF file opens and page views via Google Analytics and GTM We will provide the following: - Detailed Project Specification Document - Sample PDF files with embedded links - FTP Access to Dev Environment

$41 (Avg Bid)
$41 Avg Bid
4 bids

...centre agents. No experience is required Requirements for the role: Fast and stable internet connection (For receiving voice calls) Computer with web browser for completing online scoring SIP Softphone (such as xLite or Talk, xLite is recommended) Working hours are 8am - 9pm UK time, Monday to Friday. We expect each operator to do a minimum

$5 / hr (Avg Bid)
$5 / hr Avg Bid
39 bids

Im looking for someone to modify an open source app (linphone) to connect to our servers via API. We have our own phone servers, so we would like for the app to use our API for user login and once logged in, pull the user SIP device via API. The call log and voice mail will also pull from our API. Being the entire app is already built, it should be

$205 (Avg Bid)
$205 Avg Bid
29 bids

Configure Polycom Soundpoint ip601 SIP phones with SIPCITY cloud PBX. As above. We have three phones to be configured. This should be quick and easy for the right person.

$35 (Avg Bid)
$35 Avg Bid
2 bids
Opensip/cgrates 7 days left

Hello, I need to have an opensips server installed with cgrates. This will be for SIP trunking with nat support. I will need to be able to route inbound DIDs to customers. Also will need to be able to setup accounts in cgrates so that we can have about prepaid calling on the account. I think we have several other project that you might be able

$500 (Avg Bid)
$500 Avg Bid
1 bids

...the job was not 100% completed. I need to: 1- fix / change the logo and 2- turn off access to settings , 3- turn off the sip address when dialing a number, and 4- make the app recognize only the last 5 digits of the phone number. 5- Publish to google play. 6- give me the final source code. 1- You can download the apk file from http://www

$63 (Avg Bid)
$63 Avg Bid
12 bids
Expert in WEB-RTC -- 2 4 days left
VERIFIED

Experto en WEBRTC Tenemos un servidor asterisk y un servidor web queremos hacerlo correr en HTML5 con un cliente que tenemos hecho. You have to be a Expert in WEB-RTC we need to connect an app to asterisk via sip we already have a client and server we need to get alive.

$6 / hr (Avg Bid)
$6 / hr Avg Bid
9 bids

Hello, I would like a custom add-in to remove embedded images from email signatures in incoming email. I use MS Outlook 2016 with email service from Intermedia Exchange. The add-in should check incoming email and remove named embedded .jpg or .png images named "image00#.jgp" or "image00#.png" (for example, [url removed, login to view], [url removed, login to view]...

$187 (Avg Bid)
$187 Avg Bid
5 bids

Necesito crear un softphone que trabaje bajo protocolo sip para telefonia ip, con un proxy predefinido fijo, donde el cliente baje el softphone ojala de google play, y solo ingrese su usser (la pass sera la misma que el usser) y el proxy siempre sera el mismo, tenga la opcion de registrar la cuenta, y poder llamar.

$232 (Avg Bid)
$232 Avg Bid
17 bids

I want a basic Wordpress template that...capable of taking posts from guest users by including their email and a live-editor mode where they can embed YouTube videos by adding links to the videos and images can be embedded and the user is able to see the live preview. The submit button should make the post get submitted in the "pending posts" list.

$28 (Avg Bid)
$28 Avg Bid
11 bids
Expert in WEB-RTC 2 days left
VERIFIED

Experto en WEBRTC Tenemos un servidor asterisk y un servidor web queremos hacerlo correr en HTML5 con un cliente que tenemos hecho. You have to be a Expert in WEB-RTC we need to connect an app to asterisk via sip we already have a client and server we need to get alive.

$2 - $8 / hr
$2 - $8 / hr
0 bids

Existing jsSIP dialer needs guru troubleshooter to help resolve bug. Upon a successful connected call from web app through fpbx to sip trunk using webrtc jsSIP. We have a lag upon connection where no sound between caller and called. The lag/no sound lasts 5 seconds usually then both parties on call can hear each other. We require all work be

$204 (Avg Bid)
$204 Avg Bid
10 bids

We have outbound dialer, with FPBX to SIP trunk working. Just small delay on some #'s, not all calls, when connects until each person can hear each other. Work MUST be done via AnyDesk or TeamViewer. Here is where the delay occurs in CLI: (the lag is between these two lines) Sequence: 1. Line Posts in CLI: 0xb7503018 -- Probation passed -

$181 (Avg Bid)
$181 Avg Bid
8 bids

...install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you not robot please add "Swh" in your bid. Thanks for your attention

$206 (Avg Bid)
$206 Avg Bid
9 bids

I am using Express Talk VoIP Softphone in my Windows PC (WIN7 OS)But the Sip is not connecting properly all time Need Guidence to connect quicky all time to sip account

$5 / hr (Avg Bid)
$5 / hr Avg Bid
10 bids

...linphone. the job was not 100% completed. I need to: 1- fix the logo and 2- turn off access to settings , 3- turn off the sip address when dialing a number, and 4- make the app recognize only the last 5 digits of the phone number. 5- Publish to google play. I only have the app no source code. You can download the app from [url removed, login to view]

$94 (Avg Bid)
$94 Avg Bid
15 bids