Open source sip outbound proxy jobs

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15,499 open source sip outbound proxy jobs found, pricing in USD

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

$638 (Avg Bid)
NDA
$638 Avg Bid
2 bids
outbound sales representative 6 days left
VERIFIED

I need some help with outbound sales for a couple of my companies. You MUST have great English and understanding of Canadian / USA slang. I will need a sample of your work either from a recoding or from a voice message. This will be a per hour with large bonuses per sale!

$18 (Avg Bid)
$18 Avg Bid
8 bids

...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 of 4 hours per day.

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

I need someone who can take a proxy interview on WebSphere/Tomcat.

$555 (Avg Bid)
$555 Avg Bid
1 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 8 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

...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 [url removed, login to view] 2- Source code: [url removed, login to view]

$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 clie...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
8 bids

Greetings, Most of the way through rebuilding some aging server infrastructure, but could use some help finishing the job. Applicants have expert knowledge and experience in listed technologies and be willing to work with developer over Skype to check work / config / setup Thanks!

$23 / hr (Avg Bid)
$23 / hr Avg Bid
14 bids

This would be to create a web app for an internal single purpose web server that would process web requests from vb6 internal application, and then generate a corresponding web request for ESRI's World Geocoding Service similar too [url removed, login to view] Stree%20Dr,%20Some City%20Some State%20Some Zip, passing the result back to the orignal vb6 request

$499 (Avg Bid)
$499 Avg Bid
23 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

Client is in need of individuals from Philippines to do Outbound USA calls to sell health products.

$26666 (Avg Bid)
$26666 Avg Bid
4 bids
outbound calls 4 days left

I would like to make outbound calls to homeowners in my local area to ask about storm/hail damage to their home. If hail/storm damage is reported find out what 2 times work best for them and we will call back to confirm one of their 2 times for us to come take a look. If there is no storm/hail damage ask; "Do you have any projects on your to do list

$486 (Avg Bid)
$486 Avg Bid
20 bids

Hi I have a website that sells templates and the live previews are displayed in an iframe. SOme of these templates are from Template Monster and they contain links to the template monster website. I want those links to contain my affiliate ID so I get commission on them. Example: [url removed, login to view] As you can see it has a button that says 'Purchase Everest Now' and ...

$84 (Avg Bid)
$84 Avg Bid
7 bids
Expert in WEB-RTC 3 days left
VERIFIED

Experto en WEBRTC Tenemos un servidor asterisk y un servidor web queremos hacerlo correr en HTML5 con un clie...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