Free mobile sip dailler jobs

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7,574 free mobile sip dailler jobs found, pricing in USD

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.

$8 - $24
$8 - $24
0 bids
Opensip/cgrates 9 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.

$65 (Avg Bid)
$65 Avg Bid
9 bids
Expert in WEB-RTC -- 2 6 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

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 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.

$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
9 bids

...have a custom andriod voip client done using 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

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

...CISCO VG224 en SIP contra un Server Asterisk. Actualmente lo tenemos configurado pero con algunos problemas como que solo algunos puertos se registran y otro no. /////////////////////////////////////////////////////////////////////////////// We are looking for someone who has the ability to configure a CISCO VG224 gateway using SIP with an Asterisk

$40 (Avg Bid)
$40 Avg Bid
6 bids

Montar un PBX en Amazon. Ramales internos líneas SIP Salas de conferencias telefónicas Menú inicial Contestadoras para cada ramal I need to set up a FreePBX in an Amazon EC2 instance Set SIP lines. Internal extensions for my team (13) I need conference call rooms An initial menu Voice mail boxes por each extension

$298 (Avg Bid)
$298 Avg Bid
2 bids

...strength) of each channel. -The duration of the current call. -Activity and statistics for each SIM-card. -Rapid notification of problems via Skype, SMS or email. 5: Built-in SIP server to work with SIM card bank. We can show you the cases of the similar services which work with GoIP equipment so you can get better idea....

$2683 (Avg Bid)
$2683 Avg Bid
39 bids

I need you to develop some soft...Would need to be able to host up to 35 users on a single meeting and many meetings concurrently. I am open to the use of function Technology like web RTC or turn servers/sip. Please provide your recommended architecture and examples of your past video conferencing platform development in your response to this post.

$1244 (Avg Bid)
$1244 Avg Bid
29 bids
VOIP VPN SNOM Phone 710 2 days left
VERIFIED

I need someone to configure a SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working a...SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working and I have other devices connecting remotely. If you can config the sip settings that is an added bonus.

$37 (Avg Bid)
$37 Avg Bid
4 bids

...wholesale clients, - Setup and updates rates - Configure Plan Packs - Termination provider configuration - Setup for SMS - Configure our dialing plan to route calls to your SIP gateway: - Create wholesale clients, from whom you will be receiving calls: - Setup IP PBX management - Migrate clients from current setup a2billing/Freepbx - Secure windows

$595 (Avg Bid)
$595 Avg Bid
33 bids

...I need a FreeSWITCH (FS) configuration for the following functionality: 1. Interconnect internal endpoints 2. Link with an existing Asterisk PBX 3. Link with an external SIP trunk for incoming and outgoing calls 4. Configure basic CDR, voicemail, IVR and conference 5. Configure presence/BNL 6. Configure chat on Linphone endpoints (optional)

$256 (Avg Bid)
$256 Avg Bid
7 bids

...a developer that has a PPT (Walkie Talkie ) application that we can embbed into other appllications. The PTT platform must work with a Sip Server like OpenSip or Asterisk. Users can connect to server via any Mobile Internet, Wifi Internet or via Bluetooth hotspots that will be setup for users connection. This project is not a Design from scratch. It

$791 (Avg Bid)
$791 Avg Bid
35 bids

...post. I have a dedicated Linux box with Ubuntu [url removed, login to view] installed on it. Have also installed Asterisk but have not configured it. I would be looking for someone to setup the SIP information and install a soft phone and get it to dial out. Furthermore, to protect my PBX server I would want whatever possible basic safeguards to be in place (ie. changing

$159 (Avg Bid)
$159 Avg Bid
20 bids