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7,582 sip web programming jobs found, pricing in USD
open sip project 6 days left

Looking opensip expert ,A2billing .. please if you have skills then bid on this project otherwise donot waste my time...i need urgent requirement....

$30 (Avg Bid)
$30 Avg Bid
5 bids

...custom andriod voip client done using linphone. 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.

$84 (Avg Bid)
$84 Avg Bid
10 bids

Hi tektrix1, I noticed your profile and would like to offer you my project. We can discuss any details over chat. I wish to develop sip to viber gateway.

$1500 (Avg Bid)
$1500 Avg Bid
1 bids

required an Admin Web Portal to be utilized by me to manage the platform Admin Web Portal will have: 1. Global Configuration Management 2. Dashboard Show active call count Show online customer count Show total customer count Active Call Graph 3. DID Inventory Management 4. Outbound Ratecard and Rates management

$1165 (Avg Bid)
$1165 Avg Bid
62 bids

Hello, We need to develop a SIP to Viber gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Viber executables, or by using

$2359 (Avg Bid)
$2359 Avg Bid
7 bids

...custom andriod voip client done using linphone. 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.

$57 (Avg Bid)
$57 Avg Bid
11 bids

I have a home LAB setup consisting of Cisco Unified Communications Manager version 11.5 and Cisco Unified SIP Proxy. All set up and working. I have signed up for a trail account with [url removed, login to view] but I dont have the knowledge to get SIP outbound and inbound calls working with the components mentioned. If anyway has any experience of this and think

$26 - $328
Sealed
$26 - $328
3 bids

I used to run an asterisk box many years ago, but currently have problems with getting a sip peer to make a call. The sip invite is being rejected, and it must be something simple that I have failed tnlo find. Can you help?

$62 (Avg Bid)
$62 Avg Bid
2 bids
BULK IVR (BULK SIP CALL) 4 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

$988 (Avg Bid)
$988 Avg Bid
15 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

$552 (Avg Bid)
NDA
$552 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
46 bids

...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 fairly simple to make a few changes to use our

$207 (Avg Bid)
$207 Avg Bid
30 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 5 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

...custom andriod voip client done using linphone. 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.

$63 (Avg Bid)
$63 Avg Bid
12 bids
Expert in WEB-RTC -- 2 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.

$6 / hr (Avg Bid)
$6 / hr Avg Bid
9 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
Expert in WEB-RTC 14h 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

...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